From 148cf48d2b68354881066e2587e2673c91d2619a Mon Sep 17 00:00:00 2001
From: hrxi <hrrrxi@gmail.com>
Date: Sat, 28 Dec 2019 03:11:51 +0100
Subject: Add libnice and listen for direct connections in Jingle SOCKS5 (#608)

Add libnice as a plugin. If it is present, use libnice to enumerate
local IP addresses and listen on them to support direct connections for
Jingle SOCKS5.

Tested with Conversations and Gajim.

Created the nice.vapi file using
```
vapigen --library nice --pkg gio-2.0 --metadatadir metadata /usr/share/gir-1.0/Nice-0.1.gir
```
---
 cmake/FindNice.cmake              | 13 +++++++++++++
 cmake/PkgConfigWithFallback.cmake |  2 +-
 2 files changed, 14 insertions(+), 1 deletion(-)
 create mode 100644 cmake/FindNice.cmake

(limited to 'cmake')

diff --git a/cmake/FindNice.cmake b/cmake/FindNice.cmake
new file mode 100644
index 00000000..d40fc8c7
--- /dev/null
+++ b/cmake/FindNice.cmake
@@ -0,0 +1,13 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(Nice
+    PKG_CONFIG_NAME nice
+    LIB_NAMES nice
+    INCLUDE_NAMES nice.h
+    INCLUDE_DIR_SUFFIXES nice nice/include
+    DEPENDS GIO
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(Nice
+    REQUIRED_VARS Nice_LIBRARY
+    VERSION_VAR Nice_VERSION)
diff --git a/cmake/PkgConfigWithFallback.cmake b/cmake/PkgConfigWithFallback.cmake
index ea14fa23..9124bb35 100644
--- a/cmake/PkgConfigWithFallback.cmake
+++ b/cmake/PkgConfigWithFallback.cmake
@@ -10,7 +10,7 @@ function(find_pkg_config_with_fallback name)
     endif(PKG_CONFIG_FOUND)
 
     if (${name}_PKG_CONFIG_FOUND)
-        # Found via pkg-config, using it's result values
+        # Found via pkg-config, using its result values
         set(${name}_FOUND ${${name}_PKG_CONFIG_FOUND})
 
         # Try to find real file name of libraries
-- 
cgit v1.2.3-70-g09d2


From e6a933ad307116952d3202c36d0a8d6e7f4b0946 Mon Sep 17 00:00:00 2001
From: Marvin W <git@larma.de>
Date: Sun, 21 Mar 2021 12:41:36 +0100
Subject: Add gstreamer .cmake instructions

---
 cmake/FindGst.cmake      | 12 ++++++++++++
 cmake/FindGstApp.cmake   | 14 ++++++++++++++
 cmake/FindGstVideo.cmake | 14 ++++++++++++++
 3 files changed, 40 insertions(+)
 create mode 100644 cmake/FindGst.cmake
 create mode 100644 cmake/FindGstApp.cmake
 create mode 100644 cmake/FindGstVideo.cmake

(limited to 'cmake')

diff --git a/cmake/FindGst.cmake b/cmake/FindGst.cmake
new file mode 100644
index 00000000..942d0129
--- /dev/null
+++ b/cmake/FindGst.cmake
@@ -0,0 +1,12 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(Gst
+    PKG_CONFIG_NAME gstreamer-1.0
+    LIB_NAMES gstreamer-1.0
+    INCLUDE_NAMES gst/gst.h
+    INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(Gst
+    REQUIRED_VARS Gst_LIBRARY
+    VERSION_VAR Gst_VERSION)
diff --git a/cmake/FindGstApp.cmake b/cmake/FindGstApp.cmake
new file mode 100644
index 00000000..834b8e8e
--- /dev/null
+++ b/cmake/FindGstApp.cmake
@@ -0,0 +1,14 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GstApp
+    PKG_CONFIG_NAME gstreamer-app-1.0
+    LIB_NAMES gstapp
+    LIB_DIR_HINTS gstreamer-1.0
+    INCLUDE_NAMES gst/app/app.h
+    INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include gstreamer-app-1.0 gstreamer-app-1.0/include
+    DEPENDS Gst
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstApp
+    REQUIRED_VARS GstApp_LIBRARY
+    VERSION_VAR GstApp_VERSION)
diff --git a/cmake/FindGstVideo.cmake b/cmake/FindGstVideo.cmake
new file mode 100644
index 00000000..7d529391
--- /dev/null
+++ b/cmake/FindGstVideo.cmake
@@ -0,0 +1,14 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GstVideo
+    PKG_CONFIG_NAME gstreamer-video-1.0
+    LIB_NAMES gstvideo
+    LIB_DIR_HINTS gstreamer-1.0
+    INCLUDE_NAMES gst/video/video.h
+    INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include gstreamer-video-1.0 gstreamer-video-1.0/include
+    DEPENDS Gst
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstVideo
+    REQUIRED_VARS GstVideo_LIBRARY
+    VERSION_VAR GstVideo_VERSION)
-- 
cgit v1.2.3-70-g09d2


From ec35f95e13f4f2f756c81a35ded0980245acc5f4 Mon Sep 17 00:00:00 2001
From: fiaxh <git@lightrise.org>
Date: Wed, 24 Mar 2021 14:12:42 +0100
Subject: Add initial support for DTLS-SRTP

---
 cmake/FindGnuTLS.cmake                             |  13 +
 libdino/src/service/calls.vala                     |  22 +-
 plugins/ice/CMakeLists.txt                         |   7 +-
 plugins/ice/src/dtls_srtp.vala                     | 247 ++++++++++++
 plugins/ice/src/transport_parameters.vala          |  48 ++-
 plugins/ice/vapi/gnutls.vapi                       | 419 +++++++++++++++++++++
 plugins/rtp/CMakeLists.txt                         |   1 +
 .../src/module/xep/0166_jingle/reason_element.vala |   1 +
 xmpp-vala/src/module/xep/0166_jingle/session.vala  |  38 +-
 .../xep/0167_jingle_rtp/content_parameters.vala    |   5 +-
 .../xep/0167_jingle_rtp/jingle_rtp_module.vala     |   2 +-
 .../xep/0167_jingle_rtp/session_info_type.vala     |   2 +-
 .../0176_jingle_ice_udp/jingle_ice_udp_module.vala |   1 +
 .../0176_jingle_ice_udp/transport_parameters.vala  |  27 ++
 14 files changed, 791 insertions(+), 42 deletions(-)
 create mode 100644 cmake/FindGnuTLS.cmake
 create mode 100644 plugins/ice/src/dtls_srtp.vala
 create mode 100644 plugins/ice/vapi/gnutls.vapi

(limited to 'cmake')

diff --git a/cmake/FindGnuTLS.cmake b/cmake/FindGnuTLS.cmake
new file mode 100644
index 00000000..6b27abd7
--- /dev/null
+++ b/cmake/FindGnuTLS.cmake
@@ -0,0 +1,13 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GnuTLS
+    PKG_CONFIG_NAME gnutls
+    LIB_NAMES gnutls
+    INCLUDE_NAMES gnutls/gnutls.h
+    INCLUDE_DIR_SUFFIXES gnutls gnutls/include
+    DEPENDS GLib
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GnuTLS
+    REQUIRED_VARS GnuTLS_LIBRARY
+    VERSION_VAR GnuTLS_VERSION)
\ No newline at end of file
diff --git a/libdino/src/service/calls.vala b/libdino/src/service/calls.vala
index 5224bdd1..54c353b0 100644
--- a/libdino/src/service/calls.vala
+++ b/libdino/src/service/calls.vala
@@ -125,7 +125,7 @@ namespace Dino {
             call.state = Call.State.ESTABLISHING;
 
             if (sessions.has_key(call)) {
-                foreach (Xep.Jingle.Content content in sessions[call].contents.values) {
+                foreach (Xep.Jingle.Content content in sessions[call].contents) {
                     content.accept();
                 }
             } else {
@@ -146,7 +146,7 @@ namespace Dino {
             call.state = Call.State.DECLINED;
 
             if (sessions.has_key(call)) {
-                foreach (Xep.Jingle.Content content in sessions[call].contents.values) {
+                foreach (Xep.Jingle.Content content in sessions[call].contents) {
                     content.reject();
                 }
                 remove_call_from_datastructures(call);
@@ -223,16 +223,6 @@ namespace Dino {
             foreach (Jid full_jid in full_jids) {
                 bool supports_rtc = yield stream.get_module(Xep.JingleRtp.Module.IDENTITY).is_available(stream, full_jid);
                 if (!supports_rtc) continue;
-
-                // dtls support indicates webRTC support. Clients tend to not do normal ice udp in that case. Except Dino.
-                bool supports_dtls = yield stream_interactor.get_module(EntityInfo.IDENTITY).has_feature(conversation.account, full_jid, "urn:xmpp:jingle:apps:dtls:0");
-                if (supports_dtls) {
-                    Xep.ServiceDiscovery.Identity? identity = yield stream_interactor.get_module(EntityInfo.IDENTITY).get_identity(conversation.account, full_jid);
-                    bool is_dino = identity != null && identity.name == "Dino";
-
-                    if (!is_dino) continue;
-                }
-
                 ret.add(full_jid);
             }
             return ret;
@@ -253,7 +243,7 @@ namespace Dino {
 
         private void on_incoming_call(Account account, Xep.Jingle.Session session) {
             bool counterpart_wants_video = false;
-            foreach (Xep.Jingle.Content content in session.contents.values) {
+            foreach (Xep.Jingle.Content content in session.contents) {
                 Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
                 if (rtp_content_parameter == null) continue;
                 if (rtp_content_parameter.media == "video" && session.senders_include_us(content.senders)) {
@@ -391,7 +381,7 @@ namespace Dino {
                 on_incoming_content_add(stream, call, session, content)
             );
 
-            foreach (Xep.Jingle.Content content in session.contents.values) {
+            foreach (Xep.Jingle.Content content in session.contents) {
                 Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
                 if (rtp_content_parameter == null) continue;
 
@@ -446,7 +436,7 @@ namespace Dino {
 
             Xep.Jingle.Module jingle_module = stream_interactor.module_manager.get_module(account, Xep.Jingle.Module.IDENTITY);
             jingle_module.session_initiate_received.connect((stream, session) => {
-                foreach (Xep.Jingle.Content content in session.contents.values) {
+                foreach (Xep.Jingle.Content content in session.contents) {
                     Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
                     if (rtp_content_parameter != null) {
                         on_incoming_call(account, session);
@@ -460,7 +450,7 @@ namespace Dino {
                 if (!call_by_sid[account].has_key(session.sid)) return;
                 Call call = call_by_sid[account][session.sid];
 
-                foreach (Xep.Jingle.Content content in session.contents.values) {
+                foreach (Xep.Jingle.Content content in session.contents) {
                     if (name == null || content.content_name == name) {
                         Xep.JingleRtp.Parameters? rtp_content_parameter = content.content_params as Xep.JingleRtp.Parameters;
                         if (rtp_content_parameter != null) {
diff --git a/plugins/ice/CMakeLists.txt b/plugins/ice/CMakeLists.txt
index 90fe5b7d..38025aa0 100644
--- a/plugins/ice/CMakeLists.txt
+++ b/plugins/ice/CMakeLists.txt
@@ -2,6 +2,7 @@ find_packages(ICE_PACKAGES REQUIRED
     Gee
     GLib
     GModule
+    GnuTLS
     GObject
     GTK3
     Nice
@@ -9,8 +10,9 @@ find_packages(ICE_PACKAGES REQUIRED
 
 vala_precompile(ICE_VALA_C
 SOURCES
-    src/plugin.vala
+    src/dtls_srtp.vala
     src/module.vala
+    src/plugin.vala
     src/transport_parameters.vala
     src/util.vala
     src/register_plugin.vala
@@ -18,6 +20,7 @@ CUSTOM_VAPIS
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/dino.vapi
     ${CMAKE_BINARY_DIR}/exports/qlite.vapi
+    ${CMAKE_BINARY_DIR}/exports/crypto.vapi
 PACKAGES
     ${ICE_PACKAGES}
 OPTIONS
@@ -26,7 +29,7 @@ OPTIONS
 
 add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="ice")
 add_library(ice SHARED ${ICE_VALA_C})
-target_link_libraries(ice libdino ${ICE_PACKAGES})
+target_link_libraries(ice libdino crypto-vala ${ICE_PACKAGES})
 set_target_properties(ice PROPERTIES PREFIX "")
 set_target_properties(ice PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
 
diff --git a/plugins/ice/src/dtls_srtp.vala b/plugins/ice/src/dtls_srtp.vala
new file mode 100644
index 00000000..a21c242b
--- /dev/null
+++ b/plugins/ice/src/dtls_srtp.vala
@@ -0,0 +1,247 @@
+using GnuTLS;
+
+public class DtlsSrtp {
+
+    public signal void send_data(uint8[] data);
+
+    private X509.Certificate[] own_cert;
+    private X509.PrivateKey private_key;
+    private Cond buffer_cond = new Cond();
+    private Mutex buffer_mutex = new Mutex();
+    private Gee.LinkedList<Bytes> buffer_queue = new Gee.LinkedList<Bytes>();
+    private uint pull_timeout = uint.MAX;
+    private string peer_fingerprint;
+
+    private Crypto.Srtp.Session encrypt_session;
+    private Crypto.Srtp.Session decrypt_session;
+
+    public static DtlsSrtp setup() throws GLib.Error {
+        var obj = new DtlsSrtp();
+        obj.generate_credentials();
+        return obj;
+    }
+
+    internal string get_own_fingerprint(DigestAlgorithm digest_algo) {
+        return format_certificate(own_cert[0], digest_algo);
+    }
+
+    public void set_peer_fingerprint(string fingerprint) {
+        this.peer_fingerprint = fingerprint;
+    }
+
+    public uint8[] process_incoming_data(uint component_id, uint8[] data) {
+        if (decrypt_session != null) {
+            if (component_id == 1) return decrypt_session.decrypt_rtp(data);
+            if (component_id == 2) return decrypt_session.decrypt_rtcp(data);
+        } else if (component_id == 1) {
+            on_data_rec(data);
+        }
+        return null;
+    }
+
+    public uint8[] process_outgoing_data(uint component_id, uint8[] data) {
+        if (encrypt_session != null) {
+            if (component_id == 1) return encrypt_session.encrypt_rtp(data);
+            if (component_id == 2) return encrypt_session.encrypt_rtcp(data);
+        }
+        return null;
+    }
+
+    public void on_data_rec(owned uint8[] data) {
+        buffer_mutex.lock();
+        buffer_queue.add(new Bytes.take(data));
+        buffer_cond.signal();
+        buffer_mutex.unlock();
+    }
+
+    private void generate_credentials() throws GLib.Error {
+        int err = 0;
+
+        private_key = X509.PrivateKey.create();
+        err = private_key.generate(PKAlgorithm.RSA, 2048);
+        throw_if_error(err);
+
+        var start_time = new DateTime.now_local().add_days(1);
+        var end_time = start_time.add_days(2);
+
+        X509.Certificate cert = X509.Certificate.create();
+        cert.set_key(private_key);
+        cert.set_version(1);
+        cert.set_activation_time ((time_t) start_time.to_unix ());
+        cert.set_expiration_time ((time_t) end_time.to_unix ());
+
+        uint32 serial = 1;
+        cert.set_serial(&serial, sizeof(uint32));
+
+        cert.sign(cert, private_key);
+
+        own_cert = new X509.Certificate[] { (owned)cert };
+    }
+
+    public async void setup_dtls_connection(bool server) {
+        InitFlags server_or_client = server ? InitFlags.SERVER : InitFlags.CLIENT;
+        debug("Setting up DTLS connection. We're %s", server_or_client.to_string());
+
+        CertificateCredentials cert_cred = CertificateCredentials.create();
+        int err = cert_cred.set_x509_key(own_cert, private_key);
+        throw_if_error(err);
+
+        Session? session = Session.create(server_or_client | InitFlags.DATAGRAM);
+        session.enable_heartbeat(1);
+        session.set_srtp_profile_direct("SRTP_AES128_CM_HMAC_SHA1_80");
+        session.set_credentials(GnuTLS.CredentialsType.CERTIFICATE, cert_cred);
+        session.server_set_request(CertificateRequest.REQUEST);
+        session.set_priority_from_string("NORMAL:!VERS-TLS-ALL:+VERS-DTLS-ALL:+CTYPE-CLI-X509");
+
+        session.set_transport_pointer(this);
+        session.set_pull_function(pull_function);
+        session.set_pull_timeout_function(pull_timeout_function);
+        session.set_push_function(push_function);
+        session.set_verify_function(verify_function);
+
+        Thread<int> thread = new Thread<int> (null, () => {
+            DateTime maximum_time = new DateTime.now_utc().add_seconds(20);
+            do {
+                err = session.handshake();
+
+                DateTime current_time = new DateTime.now_utc();
+                if (maximum_time.compare(current_time) < 0) {
+                    warning("DTLS handshake timeouted");
+                    return -1;
+                }
+            } while (err < 0 && !((ErrorCode)err).is_fatal());
+            Idle.add(setup_dtls_connection.callback);
+            return err;
+        });
+        yield;
+        err = thread.join();
+
+        uint8[] km = new uint8[150];
+        Datum? client_key, client_salt, server_key, server_salt;
+        session.get_srtp_keys(km, km.length, out client_key, out client_salt, out server_key, out server_salt);
+        if (client_key == null || client_salt == null || server_key == null || server_salt == null) {
+            warning("SRTP client/server key/salt null");
+        }
+
+        Crypto.Srtp.Session encrypt_session = new Crypto.Srtp.Session(Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, 10, Crypto.Srtp.Prf.AES_CM, 0);
+        Crypto.Srtp.Session decrypt_session = new Crypto.Srtp.Session(Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, 10, Crypto.Srtp.Prf.AES_CM, 0);
+
+        if (server) {
+            encrypt_session.setkey(server_key.extract(), server_salt.extract());
+            decrypt_session.setkey(client_key.extract(), client_salt.extract());
+        } else {
+            encrypt_session.setkey(client_key.extract(), client_salt.extract());
+            decrypt_session.setkey(server_key.extract(), server_salt.extract());
+        }
+
+        this.encrypt_session = (owned)encrypt_session;
+        this.decrypt_session = (owned)decrypt_session;
+    }
+
+    private static ssize_t pull_function(void* transport_ptr, uint8[] buffer) {
+        DtlsSrtp self = transport_ptr as DtlsSrtp;
+
+        self.buffer_mutex.lock();
+        while (self.buffer_queue.size == 0) {
+            self.buffer_cond.wait(self.buffer_mutex);
+        }
+        owned Bytes data = self.buffer_queue.remove_at(0);
+        self.buffer_mutex.unlock();
+
+        uint8[] data_uint8 = Bytes.unref_to_data(data);
+        Memory.copy(buffer, data_uint8, data_uint8.length);
+
+        // The callback should return 0 on connection termination, a positive number indicating the number of bytes received, and -1 on error.
+        return (ssize_t)data.length;
+    }
+
+    private static int pull_timeout_function(void* transport_ptr, uint ms) {
+        DtlsSrtp self = transport_ptr as DtlsSrtp;
+
+        DateTime current_time = new DateTime.now_utc();
+        current_time.add_seconds(ms/1000);
+        int64 end_time = current_time.to_unix();
+
+        self.buffer_mutex.lock();
+        while (self.buffer_queue.size == 0) {
+            self.buffer_cond.wait_until(self.buffer_mutex, end_time);
+
+            DateTime new_current_time = new DateTime.now_utc();
+            if (new_current_time.compare(current_time) > 0) {
+                break;
+            }
+        }
+        self.buffer_mutex.unlock();
+
+        // The callback should return 0 on timeout, a positive number if data can be received, and -1 on error.
+        return 1;
+    }
+
+    private static ssize_t push_function(void* transport_ptr, uint8[] buffer) {
+        DtlsSrtp self = transport_ptr as DtlsSrtp;
+        self.send_data(buffer);
+
+        // The callback should return a positive number indicating the bytes sent, and -1 on error.
+        return (ssize_t)buffer.length;
+    }
+
+    private static int verify_function(Session session) {
+        DtlsSrtp self = session.get_transport_pointer() as DtlsSrtp;
+        try {
+            bool valid = self.verify_peer_cert(session);
+            if (!valid) {
+                warning("DTLS certificate invalid. Aborting handshake.");
+                return 1;
+            }
+        } catch (Error e) {
+            warning("Error during DTLS certificate validation: %s. Aborting handshake.", e.message);
+            return 1;
+        }
+
+        // The callback function should return 0 for the handshake to continue or non-zero to terminate.
+        return 0;
+    }
+
+    private bool verify_peer_cert(Session session) throws GLib.Error {
+        unowned Datum[] cert_datums = session.get_peer_certificates();
+        if (cert_datums.length == 0) {
+            warning("No peer certs");
+            return false;
+        }
+        if (cert_datums.length > 1) warning("More than one peer cert");
+
+        X509.Certificate peer_cert = X509.Certificate.create();
+        peer_cert.import(ref cert_datums[0], CertificateFormat.DER);
+
+        string peer_fp_str = format_certificate(peer_cert, DigestAlgorithm.SHA256);
+        if (peer_fp_str.down() != this.peer_fingerprint.down()) {
+            warning("First cert in peer cert list doesn't equal advertised one %s vs %s", peer_fp_str, this.peer_fingerprint);
+            return false;
+        }
+
+        return true;
+    }
+
+    private string format_certificate(X509.Certificate certificate, DigestAlgorithm digest_algo) {
+        uint8[] buf = new uint8[512];
+        size_t buf_out_size = 512;
+        certificate.get_fingerprint(digest_algo, buf, ref buf_out_size);
+
+        var sb = new StringBuilder();
+        for (int i = 0; i < buf_out_size; i++) {
+            sb.append("%02x".printf(buf[i]));
+            if (i < buf_out_size - 1) {
+                sb.append(":");
+            }
+        }
+        return sb.str;
+    }
+
+    private uint8[] uint8_pt_to_a(uint8* data, uint size) {
+        uint8[size] ret = new uint8[size];
+        for (int i = 0; i < size; i++) {
+            ret[i] = data[i];
+        }
+        return ret;
+    }
+}
\ No newline at end of file
diff --git a/plugins/ice/src/transport_parameters.vala b/plugins/ice/src/transport_parameters.vala
index a8172678..5b6431c2 100644
--- a/plugins/ice/src/transport_parameters.vala
+++ b/plugins/ice/src/transport_parameters.vala
@@ -9,9 +9,11 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
     private bool we_want_connection;
     private bool remote_credentials_set;
     private Map<uint8, DatagramConnection> connections = new HashMap<uint8, DatagramConnection>();
+    private DtlsSrtp? dtls_srtp;
 
     private class DatagramConnection : Jingle.DatagramConnection {
         private Nice.Agent agent;
+        private DtlsSrtp? dtls_srtp;
         private uint stream_id;
         private string? error;
         private ulong sent;
@@ -20,8 +22,9 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
         private ulong recv_reported;
         private ulong datagram_received_id;
 
-        public DatagramConnection(Nice.Agent agent, uint stream_id, uint8 component_id) {
+        public DatagramConnection(Nice.Agent agent, DtlsSrtp? dtls_srtp, uint stream_id, uint8 component_id) {
             this.agent = agent;
+            this.dtls_srtp = dtls_srtp;
             this.stream_id = stream_id;
             this.component_id = component_id;
             this.datagram_received_id = this.datagram_received.connect((datagram) => {
@@ -41,7 +44,12 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
 
         public override void send_datagram(Bytes datagram) {
             if (this.agent != null && is_component_ready(agent, stream_id, component_id)) {
-                agent.send(stream_id, component_id, datagram.get_data());
+                uint8[] encrypted_data = null;
+                if (dtls_srtp != null) {
+                    encrypted_data = dtls_srtp.process_outgoing_data(component_id, datagram.get_data());
+                    if (encrypted_data == null) return;
+                }
+                agent.send(stream_id, component_id, encrypted_data ?? datagram.get_data());
                 sent += datagram.length;
                 if (sent > sent_reported + 100000) {
                     debug("Sent %lu bytes via stream %u component %u", sent, stream_id, component_id);
@@ -55,6 +63,20 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
         base(components, local_full_jid, peer_full_jid, node);
         this.we_want_connection = (node == null);
         this.agent = agent;
+
+        if (this.peer_fingerprint != null || !incoming) {
+            dtls_srtp = DtlsSrtp.setup();
+            dtls_srtp.send_data.connect((data) => {
+                agent.send(stream_id, 1, data);
+            });
+            this.own_fingerprint = dtls_srtp.get_own_fingerprint(GnuTLS.DigestAlgorithm.SHA256);
+            if (incoming) {
+                dtls_srtp.set_peer_fingerprint(this.peer_fingerprint);
+            } else {
+                dtls_srtp.setup_dtls_connection(true);
+            }
+        }
+
         agent.candidate_gathering_done.connect(on_candidate_gathering_done);
         agent.initial_binding_request_received.connect(on_initial_binding_request_received);
         agent.component_state_changed.connect(on_component_state_changed);
@@ -112,6 +134,12 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
     public override void handle_transport_accept(StanzaNode transport) throws Jingle.IqError {
         debug("on_transport_accept from %s", peer_full_jid.to_string());
         base.handle_transport_accept(transport);
+
+        if (dtls_srtp != null && peer_fingerprint != null) {
+            dtls_srtp.set_peer_fingerprint(this.peer_fingerprint);
+        } else {
+            dtls_srtp = null;
+        }
     }
 
     public override void handle_transport_info(StanzaNode transport) throws Jingle.IqError {
@@ -163,9 +191,16 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
             int new_candidates = agent.set_remote_candidates(stream_id, i, candidates);
             debug("Initiated component %u with %i remote candidates", i, new_candidates);
 
-            connections[i] = new DatagramConnection(agent, stream_id, i);
+            connections[i] = new DatagramConnection(agent, dtls_srtp, stream_id, i);
             content.set_transport_connection(connections[i], i);
         }
+
+        if (incoming && dtls_srtp != null) {
+            Jingle.DatagramConnection rtp_datagram = (Jingle.DatagramConnection) content.get_transport_connection(1);
+            rtp_datagram.notify["ready"].connect(() => {
+                dtls_srtp.setup_dtls_connection(false);
+            });
+        }
         base.create_transport_connection(stream, content);
     }
 
@@ -194,12 +229,17 @@ public class Dino.Plugins.Ice.TransportParameters : JingleIceUdp.IceUdpTransport
 
     private void on_recv(Nice.Agent agent, uint stream_id, uint component_id, uint8[] data) {
         if (stream_id != this.stream_id) return;
+        uint8[] decrypt_data = null;
+        if (dtls_srtp != null) {
+            decrypt_data = dtls_srtp.process_incoming_data(component_id, data);
+            if (decrypt_data == null) return;
+        }
         may_consider_ready(stream_id, component_id);
         if (connections.has_key((uint8) component_id)) {
             if (!connections[(uint8) component_id].ready) {
                 debug("on_recv stream %u component %u when state %s", stream_id, component_id, agent.get_component_state(stream_id, component_id).to_string());
             }
-            connections[(uint8) component_id].datagram_received(new Bytes(data));
+            connections[(uint8) component_id].datagram_received(new Bytes(decrypt_data ?? data));
         } else {
             debug("on_recv stream %u component %u length %u", stream_id, component_id, data.length);
         }
diff --git a/plugins/ice/vapi/gnutls.vapi b/plugins/ice/vapi/gnutls.vapi
new file mode 100644
index 00000000..a8f75e14
--- /dev/null
+++ b/plugins/ice/vapi/gnutls.vapi
@@ -0,0 +1,419 @@
+[CCode (cprefix = "gnutls_", lower_case_cprefix = "gnutls_", cheader_filename = "gnutls/gnutls.h")]
+namespace GnuTLS {
+
+    public int global_init();
+
+    [CCode (cname = "gnutls_pull_func", has_target = false)]
+    public delegate ssize_t PullFunc(void* transport_ptr, [CCode (ctype = "void*", array_length_type="size_t")] uint8[] array);
+
+    [CCode (cname = "gnutls_pull_timeout_func", has_target = false)]
+    public delegate int PullTimeoutFunc(void* transport_ptr, uint ms);
+
+    [CCode (cname = "gnutls_push_func", has_target = false)]
+    public delegate ssize_t PushFunc(void* transport_ptr, [CCode (ctype = "void*", array_length_type="size_t")] uint8[] array);
+
+    [CCode (cname = "gnutls_certificate_verify_function", has_target = false)]
+    public delegate int VerifyFunc(Session session);
+
+    [Compact]
+    [CCode (cname = "struct gnutls_session_int", free_function = "gnutls_deinit")]
+    public class Session {
+
+        public static Session? create(int con_end) throws GLib.Error {
+            Session result;
+            var ret = init(out result, con_end);
+            throw_if_error(ret);
+            return result;
+        }
+
+        [CCode (cname = "gnutls_init")]
+        private static int init(out Session session, int con_end);
+
+        [CCode (cname = "gnutls_transport_set_push_function")]
+        public void set_push_function(PushFunc func);
+
+        [CCode (cname = "gnutls_transport_set_pull_function")]
+        public void set_pull_function(PullFunc func);
+
+        [CCode (cname = "gnutls_transport_set_pull_timeout_function")]
+        public void set_pull_timeout_function(PullTimeoutFunc func);
+
+        [CCode (cname = "gnutls_transport_set_ptr")]
+        public void set_transport_pointer(void* ptr);
+
+        [CCode (cname = "gnutls_transport_get_ptr")]
+        public void* get_transport_pointer();
+
+        [CCode (cname = "gnutls_heartbeat_enable")]
+        public int enable_heartbeat(uint type);
+
+        [CCode (cname = "gnutls_certificate_server_set_request")]
+        public void server_set_request(CertificateRequest req);
+
+        [CCode (cname = "gnutls_credentials_set")]
+        public int set_credentials_(CredentialsType type, void* cred);
+        [CCode (cname = "gnutls_credentials_set_")]
+        public void set_credentials(CredentialsType type, void* cred) throws GLib.Error {
+            int err = set_credentials_(type, cred);
+            throw_if_error(err);
+        }
+
+        [CCode (cname = "gnutls_priority_set_direct")]
+        public int set_priority_from_string_(string priority, out unowned string err_pos = null);
+        [CCode (cname = "gnutls_priority_set_direct_")]
+        public void set_priority_from_string(string priority, out unowned string err_pos = null) throws GLib.Error {
+            int err = set_priority_from_string_(priority, out err_pos);
+            throw_if_error(err);
+        }
+
+        [CCode (cname = "gnutls_srtp_set_profile_direct")]
+        public int set_srtp_profile_direct_(string profiles, out unowned string err_pos = null);
+        [CCode (cname = "gnutls_srtp_set_profile_direct_")]
+        public void set_srtp_profile_direct(string profiles, out unowned string err_pos = null) throws GLib.Error {
+            int err = set_srtp_profile_direct_(profiles, out err_pos);
+            throw_if_error(err);
+        }
+
+        [CCode (cname = "gnutls_transport_set_int")]
+        public void transport_set_int(int fd);
+
+        [CCode (cname = "gnutls_handshake")]
+        public int handshake();
+
+        [CCode (cname = "gnutls_srtp_get_keys")]
+        public int get_srtp_keys_(void *key_material, uint32 key_material_size, out Datum client_key, out Datum client_salt, out Datum server_key, out Datum server_salt);
+        [CCode (cname = "gnutls_srtp_get_keys_")]
+        public void get_srtp_keys(void *key_material, uint32 key_material_size, out Datum client_key, out Datum client_salt, out Datum server_key, out Datum server_salt) throws GLib.Error {
+            get_srtp_keys_(key_material, key_material_size, out client_key, out client_salt, out server_key, out server_salt);
+        }
+
+        [CCode (cname = "gnutls_certificate_get_peers", array_length_type = "unsigned int")]
+        public unowned Datum[]? get_peer_certificates();
+
+        [CCode (cname = "gnutls_session_set_verify_function")]
+        public void set_verify_function(VerifyFunc func);
+    }
+
+    [Compact]
+    [CCode (cname = "struct gnutls_certificate_credentials_st", free_function = "gnutls_certificate_free_credentials", cprefix = "gnutls_certificate_")]
+    public class CertificateCredentials {
+
+        [CCode (cname = "gnutls_certificate_allocate_credentials")]
+        private static int allocate(out CertificateCredentials credentials);
+
+        public static CertificateCredentials create() throws GLib.Error {
+            CertificateCredentials result;
+            var ret = allocate (out result);
+            throw_if_error(ret);
+            return result;
+        }
+
+        public void get_x509_crt(uint index, [CCode (array_length_type = "unsigned int")] out unowned X509.Certificate[] x509_ca_list);
+
+        public int set_x509_key(X509.Certificate[] cert_list, X509.PrivateKey key);
+    }
+
+    [CCode (cheader_filename = "gnutls/x509.h", cprefix = "GNUTLS_")]
+    namespace X509 {
+
+        [Compact]
+        [CCode (cname = "struct gnutls_x509_crt_int", cprefix = "gnutls_x509_crt_", free_function = "gnutls_x509_crt_deinit")]
+        public class Certificate {
+
+            [CCode (cname = "gnutls_x509_crt_init")]
+            private static int init (out Certificate cert);
+            public static Certificate create() throws GLib.Error {
+                Certificate result;
+                var ret = init (out result);
+                throw_if_error(ret);
+                return result;
+            }
+
+            [CCode (cname = "gnutls_x509_crt_import")]
+            public int import_(ref Datum data, CertificateFormat format);
+            [CCode (cname = "gnutls_x509_crt_import_")]
+            public void import(ref Datum data, CertificateFormat format) throws GLib.Error {
+                int err = import_(ref data, format);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_set_version")]
+            public int set_version_(uint version);
+            [CCode (cname = "gnutls_x509_crt_set_version_")]
+            public void set_version(uint version) throws GLib.Error {
+                int err = set_version_(version);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_set_key")]
+            public int set_key_(PrivateKey key);
+            [CCode (cname = "gnutls_x509_crt_set_key_")]
+            public void set_key(PrivateKey key) throws GLib.Error {
+                int err = set_key_(key);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_set_activation_time")]
+            public int set_activation_time_(time_t act_time);
+            [CCode (cname = "gnutls_x509_crt_set_activation_time_")]
+            public void set_activation_time(time_t act_time) throws GLib.Error {
+                int err = set_activation_time_(act_time);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_set_expiration_time")]
+            public int set_expiration_time_(time_t exp_time);
+            [CCode (cname = "gnutls_x509_crt_set_expiration_time_")]
+            public void set_expiration_time(time_t exp_time) throws GLib.Error {
+                int err = set_expiration_time_(exp_time);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_set_serial")]
+            public int set_serial_(void* serial, size_t serial_size);
+            [CCode (cname = "gnutls_x509_crt_set_serial_")]
+            public void set_serial(void* serial, size_t serial_size) throws GLib.Error {
+                int err = set_serial_(serial, serial_size);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_sign")]
+            public int sign_(Certificate issuer, PrivateKey issuer_key);
+            [CCode (cname = "gnutls_x509_crt_sign_")]
+            public void sign(Certificate issuer, PrivateKey issuer_key) throws GLib.Error {
+                int err = sign_(issuer, issuer_key);
+                throw_if_error(err);
+            }
+
+            [CCode (cname = "gnutls_x509_crt_get_fingerprint")]
+            public int get_fingerprint_(DigestAlgorithm algo, void* buf, ref size_t buf_size);
+            [CCode (cname = "gnutls_x509_crt_get_fingerprint_")]
+            public void get_fingerprint(DigestAlgorithm algo, void* buf, ref size_t buf_size) throws GLib.Error {
+                int err = get_fingerprint_(algo, buf, ref buf_size);
+                throw_if_error(err);
+            }
+        }
+
+        [Compact]
+        [CCode (cname = "struct gnutls_x509_privkey_int", cprefix = "gnutls_x509_privkey_", free_function = "gnutls_x509_privkey_deinit")]
+        public class PrivateKey {
+            private static int init (out PrivateKey key);
+            public static PrivateKey create () throws GLib.Error {
+                PrivateKey result;
+                var ret = init (out result);
+                throw_if_error(ret);
+                return result;
+            }
+
+            public int generate(PKAlgorithm algo, uint bits, uint flags = 0);
+        }
+
+    }
+
+    [CCode (cname = "gnutls_certificate_request_t", cprefix = "GNUTLS_CERT_", has_type_id = false)]
+    public enum CertificateRequest {
+        IGNORE,
+        REQUEST,
+        REQUIRE
+    }
+
+    [CCode (cname = "gnutls_pk_algorithm_t", cprefix = "GNUTLS_PK_", has_type_id = false)]
+    public enum PKAlgorithm {
+        UNKNOWN,
+        RSA,
+        DSA;
+    }
+
+    [CCode (cname = "gnutls_digest_algorithm_t", cprefix = "GNUTLS_DIG_", has_type_id = false)]
+    public enum DigestAlgorithm {
+        NULL,
+        MD5,
+        SHA1,
+        RMD160,
+        MD2,
+        SHA224,
+        SHA256,
+        SHA384,
+        SHA512;
+    }
+
+    [Flags]
+    [CCode (cname = "gnutls_init_flags_t", cprefix = "GNUTLS_", has_type_id = false)]
+    public enum InitFlags {
+        SERVER,
+        CLIENT,
+        DATAGRAM
+    }
+
+    [CCode (cname = "gnutls_credentials_type_t", cprefix = "GNUTLS_CRD_", has_type_id = false)]
+    public enum CredentialsType {
+        CERTIFICATE,
+        ANON,
+        SRP,
+        PSK,
+        IA
+    }
+
+    [CCode (cname = "gnutls_x509_crt_fmt_t", cprefix = "GNUTLS_X509_FMT_", has_type_id = false)]
+    public enum CertificateFormat {
+        DER,
+        PEM
+    }
+
+    [Flags]
+    [CCode (cname = "gnutls_certificate_status_t", cprefix = "GNUTLS_CERT_", has_type_id = false)]
+    public enum CertificateStatus {
+        INVALID,             // will be set if the certificate was not verified.
+		REVOKED,             // in X.509 this will be set only if CRLs are checked
+		SIGNER_NOT_FOUND,
+        SIGNER_NOT_CA,
+        INSECURE_ALGORITHM
+    }
+
+    [SimpleType]
+    [CCode (cname = "gnutls_datum_t", has_type_id = false)]
+    public struct Datum {
+        public uint8* data;
+        public uint size;
+
+        public uint8[] extract() {
+            uint8[size] ret = new uint8[size];
+            for (int i = 0; i < size; i++) {
+                ret[i] = data[i];
+            }
+            return ret;
+        }
+    }
+
+    // Gnutls error codes. The mapping to a TLS alert is also shown in comments.
+    [CCode (cname = "int", cprefix = "GNUTLS_E_", lower_case_cprefix = "gnutls_error_", has_type_id = false)]
+    public enum ErrorCode {
+        SUCCESS,
+        UNKNOWN_COMPRESSION_ALGORITHM,
+        UNKNOWN_CIPHER_TYPE,
+        LARGE_PACKET,
+        UNSUPPORTED_VERSION_PACKET, // GNUTLS_A_PROTOCOL_VERSION
+        UNEXPECTED_PACKET_LENGTH, // GNUTLS_A_RECORD_OVERFLOW
+        INVALID_SESSION,
+        FATAL_ALERT_RECEIVED,
+        UNEXPECTED_PACKET, // GNUTLS_A_UNEXPECTED_MESSAGE
+        WARNING_ALERT_RECEIVED,
+        ERROR_IN_FINISHED_PACKET,
+        UNEXPECTED_HANDSHAKE_PACKET,
+        UNKNOWN_CIPHER_SUITE, // GNUTLS_A_HANDSHAKE_FAILURE
+        UNWANTED_ALGORITHM,
+        MPI_SCAN_FAILED,
+        DECRYPTION_FAILED, // GNUTLS_A_DECRYPTION_FAILED, GNUTLS_A_BAD_RECORD_MAC
+        MEMORY_ERROR,
+        DECOMPRESSION_FAILED, // GNUTLS_A_DECOMPRESSION_FAILURE
+        COMPRESSION_FAILED,
+        AGAIN,
+        EXPIRED,
+        DB_ERROR,
+        SRP_PWD_ERROR,
+        INSUFFICIENT_CREDENTIALS,
+        HASH_FAILED,
+        BASE64_DECODING_ERROR,
+        MPI_PRINT_FAILED,
+        REHANDSHAKE, // GNUTLS_A_NO_RENEGOTIATION
+        GOT_APPLICATION_DATA,
+        RECORD_LIMIT_REACHED,
+        ENCRYPTION_FAILED,
+        PK_ENCRYPTION_FAILED,
+        PK_DECRYPTION_FAILED,
+        PK_SIGN_FAILED,
+        X509_UNSUPPORTED_CRITICAL_EXTENSION,
+        KEY_USAGE_VIOLATION,
+        NO_CERTIFICATE_FOUND, // GNUTLS_A_BAD_CERTIFICATE
+        INVALID_REQUEST,
+        SHORT_MEMORY_BUFFER,
+        INTERRUPTED,
+        PUSH_ERROR,
+        PULL_ERROR,
+        RECEIVED_ILLEGAL_PARAMETER, // GNUTLS_A_ILLEGAL_PARAMETER
+        REQUESTED_DATA_NOT_AVAILABLE,
+        PKCS1_WRONG_PAD,
+        RECEIVED_ILLEGAL_EXTENSION,
+        INTERNAL_ERROR,
+        DH_PRIME_UNACCEPTABLE,
+        FILE_ERROR,
+        TOO_MANY_EMPTY_PACKETS,
+        UNKNOWN_PK_ALGORITHM,
+        // returned if libextra functionality was requested but
+        // gnutls_global_init_extra() was not called.
+
+        INIT_LIBEXTRA,
+        LIBRARY_VERSION_MISMATCH,
+        // returned if you need to generate temporary RSA
+        // parameters. These are needed for export cipher suites.
+
+        NO_TEMPORARY_RSA_PARAMS,
+        LZO_INIT_FAILED,
+        NO_COMPRESSION_ALGORITHMS,
+        NO_CIPHER_SUITES,
+        OPENPGP_GETKEY_FAILED,
+        PK_SIG_VERIFY_FAILED,
+        ILLEGAL_SRP_USERNAME,
+        SRP_PWD_PARSING_ERROR,
+        NO_TEMPORARY_DH_PARAMS,
+        // For certificate and key stuff
+
+        ASN1_ELEMENT_NOT_FOUND,
+        ASN1_IDENTIFIER_NOT_FOUND,
+        ASN1_DER_ERROR,
+        ASN1_VALUE_NOT_FOUND,
+        ASN1_GENERIC_ERROR,
+        ASN1_VALUE_NOT_VALID,
+        ASN1_TAG_ERROR,
+        ASN1_TAG_IMPLICIT,
+        ASN1_TYPE_ANY_ERROR,
+        ASN1_SYNTAX_ERROR,
+        ASN1_DER_OVERFLOW,
+        OPENPGP_UID_REVOKED,
+        CERTIFICATE_ERROR,
+        CERTIFICATE_KEY_MISMATCH,
+        UNSUPPORTED_CERTIFICATE_TYPE, // GNUTLS_A_UNSUPPORTED_CERTIFICATE
+        X509_UNKNOWN_SAN,
+        OPENPGP_FINGERPRINT_UNSUPPORTED,
+        X509_UNSUPPORTED_ATTRIBUTE,
+        UNKNOWN_HASH_ALGORITHM,
+        UNKNOWN_PKCS_CONTENT_TYPE,
+        UNKNOWN_PKCS_BAG_TYPE,
+        INVALID_PASSWORD,
+        MAC_VERIFY_FAILED, // for PKCS #12 MAC
+        CONSTRAINT_ERROR,
+        WARNING_IA_IPHF_RECEIVED,
+        WARNING_IA_FPHF_RECEIVED,
+        IA_VERIFY_FAILED,
+        UNKNOWN_ALGORITHM,
+        BASE64_ENCODING_ERROR,
+        INCOMPATIBLE_CRYPTO_LIBRARY,
+        INCOMPATIBLE_LIBTASN1_LIBRARY,
+        OPENPGP_KEYRING_ERROR,
+        X509_UNSUPPORTED_OID,
+        RANDOM_FAILED,
+        BASE64_UNEXPECTED_HEADER_ERROR,
+        OPENPGP_SUBKEY_ERROR,
+        CRYPTO_ALREADY_REGISTERED,
+        HANDSHAKE_TOO_LARGE,
+        UNIMPLEMENTED_FEATURE,
+        APPLICATION_ERROR_MAX, // -65000
+        APPLICATION_ERROR_MIN; // -65500
+
+        [CCode (cname = "gnutls_error_is_fatal")]
+        public bool is_fatal();
+
+        [CCode (cname = "gnutls_perror")]
+        public void print();
+
+        [CCode (cname = "gnutls_strerror")]
+        public unowned string to_string();
+    }
+
+    public void throw_if_error(int err_int) throws GLib.Error {
+        ErrorCode error = (ErrorCode)err_int;
+        if (error != ErrorCode.SUCCESS) {
+            throw new GLib.Error(-1, error, "%s%s", error.to_string(), error.is_fatal() ? " fatal" : "");
+        }
+    }
+}
\ No newline at end of file
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 5311fac3..8ce2a7c6 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -2,6 +2,7 @@ find_packages(RTP_PACKAGES REQUIRED
     Gee
     GLib
     GModule
+    GnuTLS
     GObject
     GTK3
     Gst
diff --git a/xmpp-vala/src/module/xep/0166_jingle/reason_element.vala b/xmpp-vala/src/module/xep/0166_jingle/reason_element.vala
index 1cbdf936..4d47d4cd 100644
--- a/xmpp-vala/src/module/xep/0166_jingle/reason_element.vala
+++ b/xmpp-vala/src/module/xep/0166_jingle/reason_element.vala
@@ -24,6 +24,7 @@ namespace  Xmpp.Xep.Jingle.ReasonElement {
         BUSY,
         CANCEL,
         DECLINE,
+        GONE,
         SUCCESS
     };
 }
\ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0166_jingle/session.vala b/xmpp-vala/src/module/xep/0166_jingle/session.vala
index e9ad9169..2d359f01 100644
--- a/xmpp-vala/src/module/xep/0166_jingle/session.vala
+++ b/xmpp-vala/src/module/xep/0166_jingle/session.vala
@@ -24,9 +24,10 @@ public class Xmpp.Xep.Jingle.Session : Object {
     public Jid peer_full_jid { get; private set; }
     public bool we_initiated { get; private set; }
 
-    public HashMap<string, Content> contents = new HashMap<string, Content>();
+    public HashMap<string, Content> contents_map = new HashMap<string, Content>();
+    public Gee.List<Content> contents = new ArrayList<Content>(); // Keep the order contents
 
-    public SecurityParameters? security { get { return contents.values.to_array()[0].security_params; } }
+    public SecurityParameters? security { get { return contents.to_array()[0].security_params; } }
 
     public Session.initiate_sent(XmppStream stream, string sid, Jid local_full_jid, Jid peer_full_jid) {
         this.stream = stream;
@@ -94,7 +95,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
 
         } else if (action.has_prefix("transport-")) {
             ContentNode content_node = get_single_content_node(jingle);
-            if (!contents.has_key(content_node.name)) {
+            if (!contents_map.has_key(content_node.name)) {
                 throw new IqError.BAD_REQUEST("unknown content");
             }
 
@@ -102,7 +103,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
                 throw new IqError.BAD_REQUEST("missing transport node");
             }
 
-            Content content = contents[content_node.name];
+            Content content = contents_map[content_node.name];
 
             if (content_node.creator != content.content_creator) {
                 throw new IqError.BAD_REQUEST("unknown content; creator");
@@ -128,11 +129,11 @@ public class Xmpp.Xep.Jingle.Session : Object {
 
         } else if (action == "description-info") {
             ContentNode content_node = get_single_content_node(jingle);
-            if (!contents.has_key(content_node.name)) {
+            if (!contents_map.has_key(content_node.name)) {
                 throw new IqError.BAD_REQUEST("unknown content");
             }
 
-            Content content = contents[content_node.name];
+            Content content = contents_map[content_node.name];
 
             if (content_node.creator != content.content_creator) {
                 throw new IqError.BAD_REQUEST("unknown content; creator");
@@ -149,7 +150,8 @@ public class Xmpp.Xep.Jingle.Session : Object {
     }
 
     internal void insert_content(Content content) {
-        this.contents[content.content_name] = content;
+        this.contents_map[content.content_name] = content;
+        this.contents.add(content);
         content.set_session(this);
     }
 
@@ -209,7 +211,8 @@ public class Xmpp.Xep.Jingle.Session : Object {
 
     public async void add_content(Content content) {
         content.session = this;
-        this.contents[content.content_name] = content;
+        this.contents_map[content.content_name] = content;
+        contents.add(content);
 
         StanzaNode content_add_node = new StanzaNode.build("jingle", NS_URI)
                 .add_self_xmlns()
@@ -228,9 +231,9 @@ public class Xmpp.Xep.Jingle.Session : Object {
 
     private void handle_content_accept(ContentNode content_node) throws IqError {
         if (content_node.description == null || content_node.transport == null) throw new IqError.BAD_REQUEST("missing description or transport node");
-        if (!contents.has_key(content_node.name)) throw new IqError.BAD_REQUEST("unknown content");
+        if (!contents_map.has_key(content_node.name)) throw new IqError.BAD_REQUEST("unknown content");
 
-        Content content = contents[content_node.name];
+        Content content = contents_map[content_node.name];
 
         if (content_node.creator != content.content_creator) warning("Counterpart accepts content with an unexpected `creator`");
         if (content_node.senders != content.senders) warning("Counterpart accepts content with an unexpected `senders`");
@@ -242,7 +245,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
     private void handle_content_modify(XmppStream stream, StanzaNode jingle_node, Iq.Stanza iq) throws IqError {
         ContentNode content_node = get_single_content_node(jingle_node);
 
-        Content? content = contents[content_node.name];
+        Content? content = contents_map[content_node.name];
 
         if (content == null) throw new IqError.BAD_REQUEST("no such content");
         if (content_node.creator != content.content_creator) throw new IqError.BAD_REQUEST("mismatching creator");
@@ -301,7 +304,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
             }
         }
 
-        foreach (Content content in contents.values) {
+        foreach (Content content in contents) {
             content.terminate(false, reason_name, reason_text);
         }
 
@@ -336,7 +339,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
                 .add_self_xmlns()
                 .put_attribute("action", "session-accept")
                 .put_attribute("sid", sid);
-        foreach (Content content in contents.values) {
+        foreach (Content content in contents) {
             StanzaNode content_node = new StanzaNode.build("content", NS_URI)
                     .put_attribute("creator", "initiator")
                     .put_attribute("name", content.content_name)
@@ -345,12 +348,13 @@ public class Xmpp.Xep.Jingle.Session : Object {
                     .put_node(content.transport_params.to_transport_stanza_node());
             jingle.put_node(content_node);
         }
+
         Iq.Stanza iq = new Iq.Stanza.set(jingle) { to=peer_full_jid };
         stream.get_module(Iq.Module.IDENTITY).send_iq(stream, iq);
 
 
-        foreach (Content content in contents.values) {
-            content.on_accept(stream);
+        foreach (Content content2 in contents) {
+            content2.on_accept(stream);
         }
 
         state = State.ACTIVE;
@@ -359,7 +363,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
     internal void accept_content(Content content) {
         if (state == State.INITIATE_RECEIVED) {
             bool all_accepted = true;
-            foreach (Content c in contents.values) {
+            foreach (Content c in contents) {
                 if (c.state != Content.State.WANTS_TO_BE_ACCEPTED) {
                     all_accepted = false;
                 }
@@ -413,7 +417,7 @@ public class Xmpp.Xep.Jingle.Session : Object {
             } else {
                 reason_str = "local session-terminate";
             }
-            foreach (Content content in contents.values) {
+            foreach (Content content in contents) {
                 content.terminate(true, reason_name, reason_text);
             }
         }
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
index cca03543..32ea1df6 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
@@ -34,7 +34,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
         this.parent = parent;
         this.media = media;
         this.ssrc = ssrc;
-        this.rtcp_mux = rtcp_mux;
+        this.rtcp_mux = true;
         this.bandwidth = bandwidth;
         this.bandwidth_type = bandwidth_type;
         this.encryption_required = encryption_required;
@@ -175,6 +175,9 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
             ret.put_node(new StanzaNode.build("encryption", NS_URI)
                 .put_node(local_crypto.to_xml()));
         }
+        if (rtcp_mux) {
+            ret.put_node(new StanzaNode.build("rtcp-mux", NS_URI));
+        }
         return ret;
     }
 }
\ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
index 23aee6c9..3a9ea09f 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
@@ -84,7 +84,7 @@ public abstract class Module : XmppStreamModule {
         Jid receiver_full_jid = session.peer_full_jid;
 
         Jingle.Content? content = null;
-        foreach (Jingle.Content c in session.contents.values) {
+        foreach (Jingle.Content c in session.contents) {
             Parameters? parameters = c.content_params as Parameters;
             if (parameters == null) continue;
 
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/session_info_type.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/session_info_type.vala
index d36255f0..32cd9016 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/session_info_type.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/session_info_type.vala
@@ -50,7 +50,7 @@ namespace Xmpp.Xep.JingleRtp {
         public void send_mute(Jingle.Session session, bool mute, string media) {
             string node_name = mute ? "mute" : "unmute";
 
-            foreach (Jingle.Content content in session.contents.values) {
+            foreach (Jingle.Content content in session.contents) {
                 Parameters? parameters = content.content_params as Parameters;
                 if (parameters != null && parameters.media == media) {
                     StanzaNode session_info_content = new StanzaNode.build(node_name, NS_URI).add_self_xmlns().put_attribute("name", content.content_name);
diff --git a/xmpp-vala/src/module/xep/0176_jingle_ice_udp/jingle_ice_udp_module.vala b/xmpp-vala/src/module/xep/0176_jingle_ice_udp/jingle_ice_udp_module.vala
index 9ed494ff..4b7c7a36 100644
--- a/xmpp-vala/src/module/xep/0176_jingle_ice_udp/jingle_ice_udp_module.vala
+++ b/xmpp-vala/src/module/xep/0176_jingle_ice_udp/jingle_ice_udp_module.vala
@@ -12,6 +12,7 @@ public abstract class Module : XmppStreamModule, Jingle.Transport {
     public override void attach(XmppStream stream) {
         stream.get_module(Jingle.Module.IDENTITY).register_transport(this);
         stream.get_module(ServiceDiscovery.Module.IDENTITY).add_feature(stream, NS_URI);
+        stream.get_module(ServiceDiscovery.Module.IDENTITY).add_feature(stream, "urn:xmpp:jingle:apps:dtls:0");
     }
     public override void detach(XmppStream stream) {
         stream.get_module(ServiceDiscovery.Module.IDENTITY).remove_feature(stream, NS_URI);
diff --git a/xmpp-vala/src/module/xep/0176_jingle_ice_udp/transport_parameters.vala b/xmpp-vala/src/module/xep/0176_jingle_ice_udp/transport_parameters.vala
index 8b8aa07d..3c69d0af 100644
--- a/xmpp-vala/src/module/xep/0176_jingle_ice_udp/transport_parameters.vala
+++ b/xmpp-vala/src/module/xep/0176_jingle_ice_udp/transport_parameters.vala
@@ -13,6 +13,9 @@ public abstract class Xmpp.Xep.JingleIceUdp.IceUdpTransportParameters : Jingle.T
     public ConcurrentList<Candidate> unsent_local_candidates = new ConcurrentList<Candidate>(Candidate.equals_func);
     public Gee.List<Candidate> remote_candidates = new ArrayList<Candidate>(Candidate.equals_func);
 
+    public string? own_fingerprint = null;
+    public string? peer_fingerprint = null;
+
     public Jid local_full_jid { get; private set; }
     public Jid peer_full_jid { get; private set; }
     private uint8 components_;
@@ -34,6 +37,11 @@ public abstract class Xmpp.Xep.JingleIceUdp.IceUdpTransportParameters : Jingle.T
             foreach (StanzaNode candidateNode in node.get_subnodes("candidate")) {
                 remote_candidates.add(Candidate.parse(candidateNode));
             }
+
+            StanzaNode? fingerprint_node = node.get_subnode("fingerprint", "urn:xmpp:jingle:apps:dtls:0");
+            if (fingerprint_node != null) {
+                peer_fingerprint = fingerprint_node.get_deep_string_content();
+            }
         }
     }
 
@@ -57,6 +65,20 @@ public abstract class Xmpp.Xep.JingleIceUdp.IceUdpTransportParameters : Jingle.T
                 .add_self_xmlns()
                 .put_attribute("ufrag", local_ufrag)
                 .put_attribute("pwd", local_pwd);
+
+        if (own_fingerprint != null) {
+            var fingerprint_node = new StanzaNode.build("fingerprint", "urn:xmpp:jingle:apps:dtls:0")
+                    .add_self_xmlns()
+                    .put_attribute("hash", "sha-256")
+                    .put_node(new StanzaNode.text(own_fingerprint));
+            if (incoming) {
+                fingerprint_node.put_attribute("setup", "active");
+            } else {
+                fingerprint_node.put_attribute("setup", "actpass");
+            }
+            node.put_node(fingerprint_node);
+        }
+
         foreach (Candidate candidate in unsent_local_candidates) {
             node.put_node(candidate.to_xml());
         }
@@ -72,6 +94,11 @@ public abstract class Xmpp.Xep.JingleIceUdp.IceUdpTransportParameters : Jingle.T
         foreach (StanzaNode candidateNode in node.get_subnodes("candidate")) {
             remote_candidates.add(Candidate.parse(candidateNode));
         }
+
+        StanzaNode? fingerprint_node = node.get_subnode("fingerprint", "urn:xmpp:jingle:apps:dtls:0");
+        if (fingerprint_node != null) {
+            peer_fingerprint = fingerprint_node.get_deep_string_content();
+        }
     }
 
     public virtual void handle_transport_info(StanzaNode node) throws Jingle.IqError {
-- 
cgit v1.2.3-70-g09d2


From 5e58f2988382fffb70602cf308f6686b4731f0da Mon Sep 17 00:00:00 2001
From: Marvin W <git@larma.de>
Date: Mon, 29 Mar 2021 13:20:12 +0200
Subject: Migrate to libsrtp2

---
 cmake/FindSrtp2.cmake                  |  12 +
 plugins/crypto-vala/CMakeLists.txt     |  34 +-
 plugins/crypto-vala/src/error.vala     |   4 +-
 plugins/crypto-vala/src/srtp.c         | 836 ---------------------------------
 plugins/crypto-vala/src/srtp.h         |  82 ----
 plugins/crypto-vala/src/srtp.vala      | 122 +++++
 plugins/crypto-vala/src/srtp.vapi      | 107 -----
 plugins/crypto-vala/vapi/libsrtp2.vapi | 115 +++++
 plugins/ice/CMakeLists.txt             |   2 +-
 plugins/ice/src/dtls_srtp.vala         |  49 +-
 plugins/rtp/CMakeLists.txt             |   2 +-
 plugins/rtp/src/stream.vala            |  51 +-
 12 files changed, 314 insertions(+), 1102 deletions(-)
 create mode 100644 cmake/FindSrtp2.cmake
 delete mode 100644 plugins/crypto-vala/src/srtp.c
 delete mode 100644 plugins/crypto-vala/src/srtp.h
 create mode 100644 plugins/crypto-vala/src/srtp.vala
 delete mode 100644 plugins/crypto-vala/src/srtp.vapi
 create mode 100644 plugins/crypto-vala/vapi/libsrtp2.vapi

(limited to 'cmake')

diff --git a/cmake/FindSrtp2.cmake b/cmake/FindSrtp2.cmake
new file mode 100644
index 00000000..40b0ed97
--- /dev/null
+++ b/cmake/FindSrtp2.cmake
@@ -0,0 +1,12 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(Srtp2
+    PKG_CONFIG_NAME libsrtp2
+    LIB_NAMES srtp2
+    INCLUDE_NAMES srtp2/srtp.h
+    INCLUDE_DIR_SUFFIXES srtp2 srtp2/include
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(Srtp2
+    REQUIRED_VARS Srtp2_LIBRARY
+    VERSION_VAR Srtp2_VERSION)
\ No newline at end of file
diff --git a/plugins/crypto-vala/CMakeLists.txt b/plugins/crypto-vala/CMakeLists.txt
index f97b0d31..4a8da241 100644
--- a/plugins/crypto-vala/CMakeLists.txt
+++ b/plugins/crypto-vala/CMakeLists.txt
@@ -3,6 +3,7 @@ find_packages(CRYPTO_VALA_PACKAGES REQUIRED
     GLib
     GObject
     GIO
+    Srtp2
 )
 
 vala_precompile(CRYPTO_VALA_C
@@ -11,44 +12,23 @@ SOURCES
     "src/cipher_converter.vala"
     "src/error.vala"
     "src/random.vala"
-    "src/srtp.vapi"
+    "src/srtp.vala"
 CUSTOM_VAPIS
     "${CMAKE_CURRENT_SOURCE_DIR}/vapi/gcrypt.vapi"
+    "${CMAKE_CURRENT_SOURCE_DIR}/vapi/libsrtp2.vapi"
 PACKAGES
     ${CRYPTO_VALA_PACKAGES}
+OPTIONS
+    --vapidir=${CMAKE_CURRENT_SOURCE_DIR}/vapi
 GENERATE_VAPI
     crypto-vala
 GENERATE_HEADER
     crypto-vala
 )
 
-add_custom_command(OUTPUT "${CMAKE_BINARY_DIR}/exports/srtp.h"
-COMMAND
-    cp "${CMAKE_CURRENT_SOURCE_DIR}/src/srtp.h" "${CMAKE_BINARY_DIR}/exports/srtp.h"
-DEPENDS
-    "${CMAKE_CURRENT_SOURCE_DIR}/src/srtp.h"
-COMMENT
-    Copy header file srtp.h
-)
-
-add_custom_command(OUTPUT ${CMAKE_BINARY_DIR}/exports/crypto.vapi
-COMMAND
-    cat "${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi" "${CMAKE_CURRENT_SOURCE_DIR}/src/srtp.vapi" > "${CMAKE_BINARY_DIR}/exports/crypto.vapi"
-DEPENDS
-    ${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi
-    ${CMAKE_CURRENT_SOURCE_DIR}/src/srtp.vapi
-)
-
-add_custom_target(crypto-vapi
-DEPENDS
-    ${CMAKE_BINARY_DIR}/exports/crypto.vapi
-    ${CMAKE_BINARY_DIR}/exports/srtp.h
-)
-
-set(CFLAGS ${VALA_CFLAGS} -I${CMAKE_CURRENT_SOURCE_DIR}/src)
+set(CFLAGS ${VALA_CFLAGS})
 add_definitions(${CFLAGS})
-add_library(crypto-vala STATIC ${CRYPTO_VALA_C} src/srtp.c)
-add_dependencies(crypto-vala crypto-vapi)
+add_library(crypto-vala STATIC ${CRYPTO_VALA_C})
 target_link_libraries(crypto-vala ${CRYPTO_VALA_PACKAGES} gcrypt)
 set_property(TARGET crypto-vala PROPERTY POSITION_INDEPENDENT_CODE ON)
 
diff --git a/plugins/crypto-vala/src/error.vala b/plugins/crypto-vala/src/error.vala
index bae4ad08..5007d725 100644
--- a/plugins/crypto-vala/src/error.vala
+++ b/plugins/crypto-vala/src/error.vala
@@ -2,7 +2,9 @@ namespace Crypto {
 
 public errordomain Error {
     ILLEGAL_ARGUMENTS,
-    GCRYPT
+    GCRYPT,
+    AUTHENTICATION_FAILED,
+    UNKNOWN
 }
 
 internal void may_throw_gcrypt_error(GCrypt.Error e) throws Error {
diff --git a/plugins/crypto-vala/src/srtp.c b/plugins/crypto-vala/src/srtp.c
deleted file mode 100644
index 708244d9..00000000
--- a/plugins/crypto-vala/src/srtp.c
+++ /dev/null
@@ -1,836 +0,0 @@
-/*
- * Secure RTP with libgcrypt
- * Copyright (C) 2007  Rémi Denis-Courmont
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-/* TODO:
- * Useless stuff (because nothing depends on it):
- * - non-nul key derivation rate
- * - MKI payload
- */
-
-#ifdef HAVE_CONFIG_H
-# include <config.h>
-#endif
-
-#include <stdint.h>
-#include <stddef.h>
-
-#include "srtp.h"
-
-#include <stdbool.h>
-#include <stdlib.h>
-#include <assert.h>
-#include <errno.h>
-
-#include <gcrypt.h>
-
-#ifdef _WIN32
-# include <winsock2.h>
-#else
-# include <netinet/in.h>
-#endif
-
-#define debug( ... ) (void)0
-
-typedef struct srtp_proto_t
-{
-    gcry_cipher_hd_t cipher;
-    gcry_md_hd_t     mac;
-    uint64_t         window;
-    uint32_t         salt[4];
-} srtp_proto_t;
-
-struct srtp_session_t
-{
-    srtp_proto_t rtp;
-    srtp_proto_t rtcp;
-    unsigned flags;
-    unsigned kdr;
-    uint32_t rtcp_index;
-    uint32_t rtp_roc;
-    uint16_t rtp_seq;
-    uint16_t rtp_rcc;
-    uint8_t  tag_len;
-};
-
-enum
-{
-    SRTP_CRYPT,
-    SRTP_AUTH,
-    SRTP_SALT,
-    SRTCP_CRYPT,
-    SRTCP_AUTH,
-    SRTCP_SALT
-};
-
-
-static inline unsigned rcc_mode (const srtp_session_t *s)
-{
-    return (s->flags >> 4) & 3;
-}
-
-
-static void proto_destroy (srtp_proto_t *p)
-{
-    gcry_md_close (p->mac);
-    gcry_cipher_close (p->cipher);
-}
-
-
-/**
- * Releases all resources associated with a Secure RTP session.
- */
-void srtp_destroy (srtp_session_t *s)
-{
-    assert (s != NULL);
-
-    proto_destroy (&s->rtcp);
-    proto_destroy (&s->rtp);
-    free (s);
-}
-
-
-static int proto_create (srtp_proto_t *p, int gcipher, int gmd)
-{
-    if (gcry_cipher_open (&p->cipher, gcipher, GCRY_CIPHER_MODE_CTR, 0) == 0)
-    {
-        if (gcry_md_open (&p->mac, gmd, GCRY_MD_FLAG_HMAC) == 0)
-            return 0;
-        gcry_cipher_close (p->cipher);
-    }
-    return -1;
-}
-
-
-/**
- * Allocates a Secure RTP one-way session.
- * The same session cannot be used both ways because this would confuse
- * internal cryptographic counters; it is however of course feasible to open
- * multiple simultaneous sessions with the same master key.
- *
- * @param encr encryption algorithm number
- * @param auth authentication algortihm number
- * @param tag_len authentication tag byte length (NOT including RCC)
- * @param flags OR'ed optional flags.
- *
- * @return NULL in case of error
- */
-srtp_session_t *
-srtp_create (int encr, int auth, unsigned tag_len, int prf, unsigned flags)
-{
-    if ((flags & ~SRTP_FLAGS_MASK))
-        return NULL;
-
-    int cipher, md;
-    switch (encr)
-    {
-        case SRTP_ENCR_NULL:
-            cipher = GCRY_CIPHER_NONE;
-            break;
-
-        case SRTP_ENCR_AES_CM:
-            cipher = GCRY_CIPHER_AES;
-            break;
-
-        default:
-            return NULL;
-    }
-
-    switch (auth)
-    {
-        case SRTP_AUTH_NULL:
-            md = GCRY_MD_NONE;
-            break;
-
-        case SRTP_AUTH_HMAC_SHA1:
-            md = GCRY_MD_SHA1;
-            break;
-
-        default:
-            return NULL;
-    }
-
-    if (tag_len > gcry_md_get_algo_dlen (md))
-        return NULL;
-
-    if (prf != SRTP_PRF_AES_CM)
-        return NULL;
-
-    srtp_session_t *s = malloc (sizeof (*s));
-    if (s == NULL)
-        return NULL;
-
-    memset (s, 0, sizeof (*s));
-    s->flags = flags;
-    s->tag_len = tag_len;
-    s->rtp_rcc = 1; /* Default RCC rate */
-    if (rcc_mode (s))
-    {
-        if (tag_len < 4)
-            goto error;
-    }
-
-    if (proto_create (&s->rtp, cipher, md) == 0)
-    {
-        if (proto_create (&s->rtcp, cipher, md) == 0)
-            return s;
-        proto_destroy (&s->rtp);
-    }
-
-    error:
-    free (s);
-    return NULL;
-}
-
-
-/**
- * Counter Mode encryption/decryption (ctr length = 16 bytes)
- * with non-padded (truncated) text
- */
-static int
-do_ctr_crypt (gcry_cipher_hd_t hd, const void *ctr, uint8_t *data, size_t len)
-{
-    const size_t ctrlen = 16;
-    div_t d = div (len, ctrlen);
-
-    if (gcry_cipher_setctr (hd, ctr, ctrlen)
-        || gcry_cipher_encrypt (hd, data, d.quot * ctrlen, NULL, 0))
-        return -1;
-
-    if (d.rem)
-    {
-        /* Truncated last block */
-        uint8_t dummy[ctrlen];
-        data += d.quot * ctrlen;
-        memcpy (dummy, data, d.rem);
-        memset (dummy + d.rem, 0, ctrlen - d.rem);
-
-        if (gcry_cipher_encrypt (hd, dummy, ctrlen, data, ctrlen))
-            return -1;
-        memcpy (data, dummy, d.rem);
-    }
-
-    return 0;
-}
-
-
-/**
- * AES-CM key derivation (saltlen = 14 bytes)
- */
-static int
-do_derive (gcry_cipher_hd_t prf, const void *salt,
-           const uint8_t *r, size_t rlen, uint8_t label,
-           void *out, size_t outlen)
-{
-    uint8_t iv[16];
-
-    memcpy (iv, salt, 14);
-    iv[14] = iv[15] = 0;
-
-    assert (rlen < 14);
-    iv[13 - rlen] ^= label;
-    for (size_t i = 0; i < rlen; i++)
-        iv[sizeof (iv) - rlen + i] ^= r[i];
-
-    memset (out, 0, outlen);
-    return do_ctr_crypt (prf, iv, out, outlen);
-}
-
-
-/**
- * Sets (or resets) the master key and master salt for a SRTP session.
- * This must be done at least once before using srtp_send(), srtp_recv(),
- * srtcp_send() or srtcp_recv(). Also, rekeying is required every
- * 2^48 RTP packets or 2^31 RTCP packets (whichever comes first),
- * otherwise the protocol security might be broken.
- *
- * @return 0 on success, in case of error:
- *  EINVAL  invalid or unsupported key/salt sizes combination
- */
-int
-srtp_setkey (srtp_session_t *s, const void *key, size_t keylen,
-             const void *salt, size_t saltlen)
-{
-    /* SRTP/SRTCP cipher/salt/MAC keys derivation */
-    gcry_cipher_hd_t prf;
-    uint8_t r[6], keybuf[20];
-
-    if (saltlen != 14)
-        return EINVAL;
-
-    if (gcry_cipher_open (&prf, GCRY_CIPHER_AES, GCRY_CIPHER_MODE_CTR, 0)
-        || gcry_cipher_setkey (prf, key, keylen))
-        return EINVAL;
-
-    /* SRTP key derivation */
-#if 0
-    if (s->kdr != 0)
-    {
-        uint64_t index = (((uint64_t)s->rtp_roc) << 16) | s->rtp_seq;
-        index /= s->kdr;
-
-        for (int i = sizeof (r) - 1; i >= 0; i--)
-        {
-            r[i] = index & 0xff;
-            index = index >> 8;
-        }
-    }
-    else
-#endif
-    memset (r, 0, sizeof (r));
-    if (do_derive (prf, salt, r, 6, SRTP_CRYPT, keybuf, 16)
-        || gcry_cipher_setkey (s->rtp.cipher, keybuf, 16)
-        || do_derive (prf, salt, r, 6, SRTP_AUTH, keybuf, 20)
-        || gcry_md_setkey (s->rtp.mac, keybuf, 20)
-        || do_derive (prf, salt, r, 6, SRTP_SALT, s->rtp.salt, 14))
-        return -1;
-
-    /* SRTCP key derivation */
-    memcpy (r, &(uint32_t){ htonl (s->rtcp_index) }, 4);
-    if (do_derive (prf, salt, r, 4, SRTCP_CRYPT, keybuf, 16)
-        || gcry_cipher_setkey (s->rtcp.cipher, keybuf, 16)
-        || do_derive (prf, salt, r, 4, SRTCP_AUTH, keybuf, 20)
-        || gcry_md_setkey (s->rtcp.mac, keybuf, 20)
-        || do_derive (prf, salt, r, 4, SRTCP_SALT, s->rtcp.salt, 14))
-        return -1;
-
-    (void)gcry_cipher_close (prf);
-    return 0;
-}
-
-static int hexdigit (char c)
-{
-    if ((c >= '0') && (c <= '9'))
-        return c - '0';
-    if ((c >= 'A') && (c <= 'F'))
-        return c - 'A' + 0xA;
-    if ((c >= 'a') && (c <= 'f'))
-        return c - 'a' + 0xa;
-    return -1;
-}
-
-static ssize_t hexstring (const char *in, uint8_t *out, size_t outlen)
-{
-    size_t inlen = strlen (in);
-
-    if ((inlen > (2 * outlen)) || (inlen & 1))
-        return -1;
-
-    for (size_t i = 0; i < inlen; i += 2)
-    {
-        int a = hexdigit (in[i]), b = hexdigit (in[i + 1]);
-        if ((a == -1) || (b == -1))
-            return -1;
-        out[i / 2] = (a << 4) | b;
-    }
-    return inlen / 2;
-}
-
-/**
- * Sets (or resets) the master key and master salt for a SRTP session
- * from hexadecimal strings. See also srtp_setkey().
- *
- * @return 0 on success, in case of error:
- *  EINVAL  invalid or unsupported key/salt sizes combination
- */
-int
-srtp_setkeystring (srtp_session_t *s, const char *key, const char *salt)
-{
-    uint8_t bkey[16]; /* TODO/NOTE: hard-coded for AES */
-    uint8_t bsalt[14]; /* TODO/NOTE: hard-coded for the PRF-AES-CM */
-    ssize_t bkeylen = hexstring (key, bkey, sizeof (bkey));
-    ssize_t bsaltlen = hexstring (salt, bsalt, sizeof (bsalt));
-
-    if ((bkeylen == -1) || (bsaltlen == -1))
-        return EINVAL;
-    return srtp_setkey (s, bkey, bkeylen, bsalt, bsaltlen) ? EINVAL : 0;
-}
-
-/**
- * Sets Roll-over-Counter Carry (RCC) rate for the SRTP session. If not
- * specified (through this function), the default rate of ONE is assumed
- * (i.e. every RTP packets will carry the RoC). RCC rate is ignored if none
- * of the RCC mode has been selected.
- *
- * The RCC mode is selected through one of these flags for srtp_create():
- *  SRTP_RCC_MODE1: integrity protection only for RoC carrying packets
- *  SRTP_RCC_MODE2: integrity protection for all packets
- *  SRTP_RCC_MODE3: no integrity protection
- *
- * RCC mode 3 is insecure. Compared to plain RTP, it provides confidentiality
- * (through encryption) but is much more prone to DoS. It can only be used if
- * anti-spoofing protection is provided by lower network layers (e.g. IPsec,
- * or trusted routers and proper source address filtering).
- *
- * If RCC rate is 1, RCC mode 1 and 2 are functionally identical.
- *
- * @param rate RoC Carry rate (MUST NOT be zero)
- */
-void srtp_setrcc_rate (srtp_session_t *s, uint16_t rate)
-{
-    assert (rate != 0);
-    s->rtp_rcc = rate;
-}
-
-
-/** AES-CM for RTP (salt = 14 bytes + 2 nul bytes) */
-static int
-rtp_crypt (gcry_cipher_hd_t hd, uint32_t ssrc, uint32_t roc, uint16_t seq,
-           const uint32_t *salt, uint8_t *data, size_t len)
-{
-    /* Determines cryptographic counter (IV) */
-    uint32_t counter[4];
-    counter[0] = salt[0];
-    counter[1] = salt[1] ^ ssrc;
-    counter[2] = salt[2] ^ htonl (roc);
-    counter[3] = salt[3] ^ htonl (seq << 16);
-
-    /* Encryption */
-    return do_ctr_crypt (hd, counter, data, len);
-}
-
-
-/** Determines SRTP Roll-Over-Counter (in host-byte order) */
-static uint32_t
-srtp_compute_roc (const srtp_session_t *s, uint16_t seq)
-{
-    uint32_t roc = s->rtp_roc;
-
-    if (((seq - s->rtp_seq) & 0xffff) < 0x8000)
-    {
-        /* Sequence is ahead, good */
-        if (seq < s->rtp_seq)
-            roc++; /* Sequence number wrap */
-    }
-    else
-    {
-        /* Sequence is late, bad */
-        if (seq > s->rtp_seq)
-            roc--; /* Wrap back */
-    }
-    return roc;
-}
-
-
-/** Returns RTP sequence (in host-byte order) */
-static inline uint16_t rtp_seq (const uint8_t *buf)
-{
-    return (buf[2] << 8) | buf[3];
-}
-
-
-/** Message Authentication and Integrity for RTP */
-static const uint8_t *
-rtp_digest (gcry_md_hd_t md, const uint8_t *data, size_t len,
-            uint32_t roc)
-{
-    gcry_md_reset (md);
-    gcry_md_write (md, data, len);
-    gcry_md_write (md, &(uint32_t){ htonl (roc) }, 4);
-    return gcry_md_read (md, 0);
-}
-
-
-/**
- * Encrypts/decrypts a RTP packet and updates SRTP context
- * (CTR block cypher mode of operation has identical encryption and
- * decryption function).
- *
- * @param buf RTP packet to be en-/decrypted
- * @param len RTP packet length
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted RTP packet
- *  EACCES  replayed packet or out-of-window or sync lost
- */
-static int srtp_crypt (srtp_session_t *s, uint8_t *buf, size_t len)
-{
-    assert (s != NULL);
-    assert (len >= 12u);
-
-    if ((buf[0] >> 6) != 2)
-        return EINVAL;
-
-    /* Computes encryption offset */
-    uint16_t offset = 12;
-    offset += (buf[0] & 0xf) * 4; // skips CSRC
-
-    if (buf[0] & 0x10)
-    {
-        uint16_t extlen;
-
-        offset += 4;
-        if (len < offset)
-            return EINVAL;
-
-        memcpy (&extlen, buf + offset - 2, 2);
-        offset += htons (extlen); // skips RTP extension header
-    }
-
-    if (len < offset)
-        return EINVAL;
-
-    /* Determines RTP 48-bits counter and SSRC */
-    uint16_t seq = rtp_seq (buf);
-    uint32_t roc = srtp_compute_roc (s, seq), ssrc;
-    memcpy (&ssrc, buf + 8, 4);
-
-    /* Updates ROC and sequence (it's safe now) */
-    int16_t diff = seq - s->rtp_seq;
-    if (diff > 0)
-    {
-        /* Sequence in the future, good */
-        s->rtp.window = s->rtp.window << diff;
-        s->rtp.window |= UINT64_C(1);
-        s->rtp_seq = seq, s->rtp_roc = roc;
-    }
-    else
-    {
-        /* Sequence in the past/present, bad */
-        diff = -diff;
-        if ((diff >= 64) || ((s->rtp.window >> diff) & 1))
-            return EACCES; /* Replay attack */
-        s->rtp.window |= UINT64_C(1) << diff;
-    }
-
-    /* Encrypt/Decrypt */
-    if (s->flags & SRTP_UNENCRYPTED)
-        return 0;
-
-    if (rtp_crypt (s->rtp.cipher, ssrc, roc, seq, s->rtp.salt,
-                   buf + offset, len - offset))
-        return EINVAL;
-
-    return 0;
-}
-
-
-/**
- * Turns a RTP packet into a SRTP packet: encrypt it, then computes
- * the authentication tag and appends it.
- * Note that you can encrypt packet in disorder.
- *
- * @param buf RTP packet to be encrypted/digested
- * @param lenp pointer to the RTP packet length on entry,
- *             set to the SRTP length on exit (undefined on non-ENOSPC error)
- * @param bufsize size (bytes) of the packet buffer
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted RTP packet or internal error
- *  ENOSPC  bufsize is too small to add authentication tag
- *          (<lenp> will hold the required byte size)
- *  EACCES  packet would trigger a replay error on receiver
- */
-int
-srtp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t bufsize)
-{
-    size_t len = *lenp;
-    size_t tag_len;
-    size_t roc_len = 0;
-
-    /* Compute required buffer size */
-    if (len < 12u)
-        return EINVAL;
-
-    if (!(s->flags & SRTP_UNAUTHENTICATED))
-    {
-        tag_len = s->tag_len;
-
-        if (rcc_mode (s))
-        {
-            assert (tag_len >= 4);
-            assert (s->rtp_rcc != 0);
-            if ((rtp_seq (buf) % s->rtp_rcc) == 0)
-            {
-                roc_len = 4;
-                if (rcc_mode (s) == 3)
-                    tag_len = 0; /* RCC mode 3 -> no auth*/
-                else
-                    tag_len -= 4; /* RCC mode 1 or 2 -> auth*/
-            }
-            else
-            {
-                if (rcc_mode (s) & 1)
-                    tag_len = 0; /* RCC mode 1 or 3 -> no auth */
-            }
-        }
-
-        *lenp = len + roc_len + tag_len;
-    }
-    else
-        tag_len = 0;
-
-    if (bufsize < *lenp)
-        return ENOSPC;
-
-    /* Encrypt payload */
-    int val = srtp_crypt (s, buf, len);
-    if (val)
-        return val;
-
-    /* Authenticate payload */
-    if (!(s->flags & SRTP_UNAUTHENTICATED))
-    {
-        uint32_t roc = srtp_compute_roc (s, rtp_seq (buf));
-        const uint8_t *tag = rtp_digest (s->rtp.mac, buf, len, roc);
-
-        if (roc_len)
-        {
-            memcpy (buf + len, &(uint32_t){ htonl (s->rtp_roc) }, 4);
-            len += 4;
-        }
-        memcpy (buf + len, tag, tag_len);
-#if 0
-        printf ("Sent    : 0x");
-        for (unsigned i = 0; i < tag_len; i++)
-            printf ("%02x", tag[i]);
-        puts ("");
-#endif
-    }
-
-    return 0;
-}
-
-
-/**
- * Turns a SRTP packet into a RTP packet: authenticates the packet,
- * then decrypts it.
- *
- * @param buf RTP packet to be digested/decrypted
- * @param lenp pointer to the SRTP packet length on entry,
- *             set to the RTP length on exit (undefined in case of error)
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted SRTP packet
- *  EACCES  authentication failed (spoofed packet or out-of-sync)
- */
-int
-srtp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp)
-{
-    size_t len = *lenp;
-    if (len < 12u)
-        return EINVAL;
-
-    if (!(s->flags & SRTP_UNAUTHENTICATED))
-    {
-        size_t tag_len = s->tag_len, roc_len = 0;
-        if (rcc_mode (s))
-        {
-            if ((rtp_seq (buf) % s->rtp_rcc) == 0)
-            {
-                roc_len = 4;
-                if (rcc_mode (s) == 3)
-                    tag_len = 0;
-                else
-                    tag_len -= 4;
-            }
-            else
-            {
-                if (rcc_mode (s) & 1)
-                    tag_len = 0; // RCC mode 1 or 3: no auth
-            }
-        }
-
-        if (len < (12u + roc_len + tag_len))
-            return EINVAL;
-        len -= roc_len + tag_len;
-
-        uint32_t roc = srtp_compute_roc (s, rtp_seq (buf)), rcc;
-        if (roc_len)
-        {
-            assert (roc_len == 4);
-            memcpy (&rcc, buf + len, 4);
-            rcc = ntohl (rcc);
-        }
-        else
-            rcc = roc;
-
-        const uint8_t *tag = rtp_digest (s->rtp.mac, buf, len, rcc);
-#if 0
-        printf ("Computed: 0x");
-        for (unsigned i = 0; i < tag_len; i++)
-            printf ("%02x", tag[i]);
-        printf ("\nReceived: 0x");
-        for (unsigned i = 0; i < tag_len; i++)
-            printf ("%02x", buf[len + roc_len + i]);
-        puts ("");
-#endif
-        if (memcmp (buf + len + roc_len, tag, tag_len))
-            return EACCES;
-
-        if (roc_len)
-        {
-            /* Authenticated packet carried a Roll-Over-Counter */
-            s->rtp_roc += rcc - roc;
-            assert (srtp_compute_roc (s, rtp_seq (buf)) == rcc);
-        }
-        *lenp = len;
-    }
-
-    return srtp_crypt (s, buf, len);
-}
-
-
-/** AES-CM for RTCP (salt = 14 bytes + 2 nul bytes) */
-static int
-rtcp_crypt (gcry_cipher_hd_t hd, uint32_t ssrc, uint32_t index,
-            const uint32_t *salt, uint8_t *data, size_t len)
-{
-    return rtp_crypt (hd, ssrc, index >> 16, index & 0xffff, salt, data, len);
-}
-
-
-/** Message Authentication and Integrity for RTCP */
-static const uint8_t *
-rtcp_digest (gcry_md_hd_t md, const void *data, size_t len)
-{
-    gcry_md_reset (md);
-    gcry_md_write (md, data, len);
-    return gcry_md_read (md, 0);
-}
-
-
-/**
- * Encrypts/decrypts a RTCP packet and updates SRTCP context
- * (CTR block cypher mode of operation has identical encryption and
- * decryption function).
- *
- * @param buf RTCP packet to be en-/decrypted
- * @param len RTCP packet length
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted RTCP packet
- */
-static int srtcp_crypt (srtp_session_t *s, uint8_t *buf, size_t len)
-{
-    assert (s != NULL);
-
-    /* 8-bytes unencrypted header, and 4-bytes unencrypted footer */
-    if ((len < 12) || ((buf[0] >> 6) != 2))
-        return EINVAL;
-
-    uint32_t index;
-    memcpy (&index, buf + len, 4);
-    index = ntohl (index);
-    if (((index >> 31) != 0) != ((s->flags & SRTCP_UNENCRYPTED) == 0))
-        return EINVAL; // E-bit mismatch
-
-    index &= ~(1 << 31); // clear E-bit for counter
-
-    /* Updates SRTCP index (safe here) */
-    int32_t diff = index - s->rtcp_index;
-    if (diff > 0)
-    {
-        /* Packet in the future, good */
-        s->rtcp.window = s->rtcp.window << diff;
-        s->rtcp.window |= UINT64_C(1);
-        s->rtcp_index = index;
-    }
-    else
-    {
-        /* Packet in the past/present, bad */
-        diff = -diff;
-        if ((diff >= 64) || ((s->rtcp.window >> diff) & 1))
-            return EACCES; // replay attack!
-        s->rtp.window |= UINT64_C(1) << diff;
-    }
-
-    /* Crypts SRTCP */
-    if (s->flags & SRTCP_UNENCRYPTED)
-        return 0;
-
-    uint32_t ssrc;
-    memcpy (&ssrc, buf + 4, 4);
-
-    if (rtcp_crypt (s->rtcp.cipher, ssrc, index, s->rtp.salt,
-                    buf + 8, len - 8))
-        return EINVAL;
-    return 0;
-}
-
-
-/**
- * Turns a RTCP packet into a SRTCP packet: encrypt it, then computes
- * the authentication tag and appends it.
- *
- * @param buf RTCP packet to be encrypted/digested
- * @param lenp pointer to the RTCP packet length on entry,
- *             set to the SRTCP length on exit (undefined in case of error)
- * @param bufsize size (bytes) of the packet buffer
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted RTCP packet or internal error
- *  ENOSPC  bufsize is too small (to add index and authentication tag)
- */
-int
-srtcp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t bufsize)
-{
-    size_t len = *lenp;
-    if (bufsize < (len + 4 + s->tag_len))
-        return ENOSPC;
-
-    uint32_t index = ++s->rtcp_index;
-    if (index >> 31)
-        s->rtcp_index = index = 0; /* 31-bit wrap */
-
-    if ((s->flags & SRTCP_UNENCRYPTED) == 0)
-        index |= 0x80000000; /* Set Encrypted bit */
-    memcpy (buf + len, &(uint32_t){ htonl (index) }, 4);
-
-    int val = srtcp_crypt (s, buf, len);
-    if (val)
-        return val;
-
-    len += 4; /* Digests SRTCP index too */
-
-    const uint8_t *tag = rtcp_digest (s->rtcp.mac, buf, len);
-    memcpy (buf + len, tag, s->tag_len);
-    *lenp = len + s->tag_len;
-    return 0;
-}
-
-
-/**
- * Turns a SRTCP packet into a RTCP packet: authenticates the packet,
- * then decrypts it.
- *
- * @param buf RTCP packet to be digested/decrypted
- * @param lenp pointer to the SRTCP packet length on entry,
- *             set to the RTCP length on exit (undefined in case of error)
- *
- * @return 0 on success, in case of error:
- *  EINVAL  malformatted SRTCP packet
- *  EACCES  authentication failed (spoofed packet or out-of-sync)
- */
-int
-srtcp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp)
-{
-    size_t len = *lenp;
-
-    if (len < (4u + s->tag_len))
-        return EINVAL;
-    len -= s->tag_len;
-
-    const uint8_t *tag = rtcp_digest (s->rtcp.mac, buf, len);
-    if (memcmp (buf + len, tag, s->tag_len))
-        return EACCES;
-
-    len -= 4; /* Remove SRTCP index before decryption */
-    *lenp = len;
-    return srtcp_crypt (s, buf, len);
-}
\ No newline at end of file
diff --git a/plugins/crypto-vala/src/srtp.h b/plugins/crypto-vala/src/srtp.h
deleted file mode 100644
index abca6988..00000000
--- a/plugins/crypto-vala/src/srtp.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Secure RTP with libgcrypt
- * Copyright (C) 2007  Rémi Denis-Courmont
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1
- * of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- ****************************************************************************/
-
-#ifndef LIBVLC_SRTP_H
-# define LIBVLC_SRTP_H 1
-#include <stdint.h>
-
-typedef struct srtp_session_t srtp_session_t;
-
-enum
-{
-    SRTP_UNENCRYPTED=0x1,   //< do not encrypt SRTP packets
-    SRTCP_UNENCRYPTED=0x2,  //< do not encrypt SRTCP packets
-    SRTP_UNAUTHENTICATED=0x4, //< authenticate only SRTCP packets
-
-    SRTP_RCC_MODE1=0x10,    //< use Roll-over-Counter Carry mode 1
-    SRTP_RCC_MODE2=0x20,    //< use Roll-over-Counter Carry mode 2
-    SRTP_RCC_MODE3=0x30,    //< use Roll-over-Counter Carry mode 3 (insecure)
-
-    SRTP_FLAGS_MASK=0x37    //< mask for valid flags
-};
-
-/** SRTP encryption algorithms (ciphers); same values as MIKEY */
-enum
-{
-    SRTP_ENCR_NULL=0,   //< no encryption
-    SRTP_ENCR_AES_CM=1, //< AES counter mode
-    SRTP_ENCR_AES_F8=2, //< AES F8 mode (not implemented)
-};
-
-/** SRTP authenticaton algorithms; same values as MIKEY */
-enum
-{
-    SRTP_AUTH_NULL=0,      //< no authentication code
-    SRTP_AUTH_HMAC_SHA1=1, //< HMAC-SHA1
-};
-
-/** SRTP pseudo random function; same values as MIKEY */
-enum
-{
-    SRTP_PRF_AES_CM=0, //< AES counter mode
-};
-
-# ifdef __cplusplus
-extern "C" {
-# endif
-
-srtp_session_t *srtp_create (int encr, int auth, unsigned tag_len, int prf,
-                             unsigned flags);
-void srtp_destroy (srtp_session_t *s);
-
-int srtp_setkey (srtp_session_t *s, const void *key, size_t keylen,
-                 const void *salt, size_t saltlen);
-int srtp_setkeystring (srtp_session_t *s, const char *key, const char *salt);
-
-void srtp_setrcc_rate (srtp_session_t *s, uint16_t rate);
-
-int srtp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t maxsize);
-int srtp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp);
-int srtcp_send (srtp_session_t *s, uint8_t *buf, size_t *lenp, size_t maxsiz);
-int srtcp_recv (srtp_session_t *s, uint8_t *buf, size_t *lenp);
-
-# ifdef __cplusplus
-}
-# endif
-#endif
\ No newline at end of file
diff --git a/plugins/crypto-vala/src/srtp.vala b/plugins/crypto-vala/src/srtp.vala
new file mode 100644
index 00000000..77b5acde
--- /dev/null
+++ b/plugins/crypto-vala/src/srtp.vala
@@ -0,0 +1,122 @@
+using Srtp;
+
+public class Crypto.Srtp {
+    public const string AES_CM_128_HMAC_SHA1_80 = "AES_CM_128_HMAC_SHA1_80";
+    public const string AES_CM_128_HMAC_SHA1_32 = "AES_CM_128_HMAC_SHA1_32";
+    public const string F8_128_HMAC_SHA1_80 = "F8_128_HMAC_SHA1_80";
+
+    public class Session {
+        public bool has_encrypt { get; private set; }
+        public bool has_decrypt { get; private set; }
+
+        private Context encrypt_context;
+        private Context decrypt_context;
+
+        static construct {
+            init();
+            install_log_handler(log);
+        }
+
+        private static void log(LogLevel level, string msg) {
+            print(@"SRTP[$level]: $msg\n");
+        }
+
+        public Session() {
+            Context.create(out encrypt_context, null);
+            Context.create(out decrypt_context, null);
+        }
+
+        public uint8[] encrypt_rtp(uint8[] data) throws Error {
+            uint8[] buf = new uint8[data.length + MAX_TRAILER_LEN];
+            Memory.copy(buf, data, data.length);
+            int buf_use = data.length;
+            ErrorStatus res = encrypt_context.protect(buf, ref buf_use);
+            if (res != ErrorStatus.ok) {
+                throw new Error.UNKNOWN(@"SRTP encrypt failed: $res");
+            }
+            uint8[] ret = new uint8[buf_use];
+            GLib.Memory.copy(ret, buf, buf_use);
+            return ret;
+        }
+
+        public uint8[] decrypt_rtp(uint8[] data) throws Error {
+            uint8[] buf = new uint8[data.length];
+            Memory.copy(buf, data, data.length);
+            int buf_use = data.length;
+            ErrorStatus res = decrypt_context.unprotect(buf, ref buf_use);
+            switch (res) {
+                case ErrorStatus.auth_fail:
+                    throw new Error.AUTHENTICATION_FAILED("SRTP packet failed the message authentication check");
+                case ErrorStatus.ok:
+                    break;
+                default:
+                    throw new Error.UNKNOWN(@"SRTP decrypt failed: $res");
+            }
+            uint8[] ret = new uint8[buf_use];
+            GLib.Memory.copy(ret, buf, buf_use);
+            return ret;
+        }
+
+        public uint8[] encrypt_rtcp(uint8[] data) throws Error {
+            uint8[] buf = new uint8[data.length + MAX_TRAILER_LEN + 4];
+            Memory.copy(buf, data, data.length);
+            int buf_use = data.length;
+            ErrorStatus res = encrypt_context.protect_rtcp(buf, ref buf_use);
+            if (res != ErrorStatus.ok) {
+                throw new Error.UNKNOWN(@"SRTCP encrypt failed: $res");
+            }
+            uint8[] ret = new uint8[buf_use];
+            GLib.Memory.copy(ret, buf, buf_use);
+            return ret;
+        }
+
+        public uint8[] decrypt_rtcp(uint8[] data) throws Error {
+            uint8[] buf = new uint8[data.length];
+            Memory.copy(buf, data, data.length);
+            int buf_use = data.length;
+            ErrorStatus res = decrypt_context.unprotect_rtcp(buf, ref buf_use);
+            switch (res) {
+                case ErrorStatus.auth_fail:
+                    throw new Error.AUTHENTICATION_FAILED("SRTCP packet failed the message authentication check");
+                case ErrorStatus.ok:
+                    break;
+                default:
+                    throw new Error.UNKNOWN(@"SRTP decrypt failed: $res");
+            }
+            uint8[] ret = new uint8[buf_use];
+            GLib.Memory.copy(ret, buf, buf_use);
+            return ret;
+        }
+
+        private Policy create_policy(string profile) {
+            Policy policy = Policy();
+            switch (profile) {
+                case AES_CM_128_HMAC_SHA1_80:
+                    policy.rtp.set_aes_cm_128_hmac_sha1_80();
+                    policy.rtcp.set_aes_cm_128_hmac_sha1_80();
+                    break;
+            }
+            return policy;
+        }
+
+        public void set_encryption_key(string profile, uint8[] key, uint8[] salt) {
+            Policy policy = create_policy(profile);
+            policy.ssrc.type = SsrcType.any_outbound;
+            policy.key = new uint8[key.length + salt.length];
+            Memory.copy(policy.key, key, key.length);
+            Memory.copy(((uint8*)policy.key) + key.length, salt, salt.length);
+            encrypt_context.add_stream(ref policy);
+            has_encrypt = true;
+        }
+
+        public void set_decryption_key(string profile, uint8[] key, uint8[] salt) {
+            Policy policy = create_policy(profile);
+            policy.ssrc.type = SsrcType.any_inbound;
+            policy.key = new uint8[key.length + salt.length];
+            Memory.copy(policy.key, key, key.length);
+            Memory.copy(((uint8*)policy.key) + key.length, salt, salt.length);
+            decrypt_context.add_stream(ref policy);
+            has_decrypt = true;
+        }
+    }
+}
\ No newline at end of file
diff --git a/plugins/crypto-vala/src/srtp.vapi b/plugins/crypto-vala/src/srtp.vapi
deleted file mode 100644
index 0fe825c3..00000000
--- a/plugins/crypto-vala/src/srtp.vapi
+++ /dev/null
@@ -1,107 +0,0 @@
-[CCode (cheader_filename="srtp.h")]
-namespace Crypto.Srtp {
-
-[Compact]
-[CCode (cname = "srtp_session_t", free_function = "srtp_destroy")]
-public class Session {
-    [CCode (cname = "srtp_create")]
-    public Session(Encryption encr, Authentication auth, uint tag_len, Prf prf, Flags flags);
-    [CCode (cname = "srtp_setkey")]
-    public int setkey(uint8[] key, uint8[] salt);
-    [CCode (cname = "srtp_setkeystring")]
-    public int setkeystring(string key, string salt);
-    [CCode (cname = "srtp_setrcc_rate")]
-    public void setrcc_rate(uint16 rate);
-
-    [CCode (cname = "srtp_send")]
-    private int rtp_send([CCode (array_length = false)] uint8[] buf, ref size_t len, size_t maxsize);
-    [CCode (cname = "srtcp_send")]
-    private int rtcp_send([CCode (array_length = false)] uint8[] buf, ref size_t len, size_t maxsize);
-    [CCode (cname = "srtp_recv")]
-    private int rtp_recv([CCode (array_length = false)] uint8[] buf, ref size_t len);
-    [CCode (cname = "srtcp_recv")]
-    private int rtcp_recv([CCode (array_length = false)] uint8[] buf, ref size_t len);
-
-    public uint8[] encrypt_rtp(uint8[] input, uint tag_len = 10) throws GLib.Error {
-        uint8[] buf = new uint8[input.length + tag_len];
-        GLib.Memory.copy(buf, input, input.length);
-        size_t buf_use = input.length;
-        int res = rtp_send(buf, ref buf_use, buf.length);
-        if (res != 0) {
-            throw new GLib.Error(-1, res, "RTP encrypt failed");
-        }
-        uint8[] ret = new uint8[buf_use];
-        GLib.Memory.copy(ret, buf, buf_use);
-        return ret;
-    }
-
-    public uint8[] encrypt_rtcp(uint8[] input, uint tag_len = 10) throws GLib.Error {
-        uint8[] buf = new uint8[input.length + tag_len + 4];
-        GLib.Memory.copy(buf, input, input.length);
-        size_t buf_use = input.length;
-        int res = rtcp_send(buf, ref buf_use, buf.length);
-        if (res != 0) {
-            throw new GLib.Error(-1, res, "RTCP encrypt failed");
-        }
-        uint8[] ret = new uint8[buf_use];
-        GLib.Memory.copy(ret, buf, buf_use);
-        return ret;
-    }
-
-    public uint8[] decrypt_rtp(uint8[] input) throws GLib.Error {
-        uint8[] buf = new uint8[input.length];
-        GLib.Memory.copy(buf, input, input.length);
-        size_t buf_use = input.length;
-        int res = rtp_recv(buf, ref buf_use);
-        if (res != 0) {
-            throw new GLib.Error(-1, res, "RTP decrypt failed");
-        }
-        uint8[] ret = new uint8[buf_use];
-        GLib.Memory.copy(ret, buf, buf_use);
-        return ret;
-    }
-
-    public uint8[] decrypt_rtcp(uint8[] input) throws GLib.Error {
-        uint8[] buf = new uint8[input.length];
-        GLib.Memory.copy(buf, input, input.length);
-        size_t buf_use = input.length;
-        int res = rtcp_recv(buf, ref buf_use);
-        if (res != 0) {
-            throw new GLib.Error(-1, res, "RTCP decrypt failed");
-        }
-        uint8[] ret = new uint8[buf_use];
-        GLib.Memory.copy(ret, buf, buf_use);
-        return ret;
-    }
-}
-
-[Flags]
-[CCode (cname = "unsigned", cprefix = "", has_type_id = false)]
-public enum Flags {
-    SRTP_UNENCRYPTED,
-    SRTCP_UNENCRYPTED,
-    SRTP_UNAUTHENTICATED,
-    SRTP_RCC_MODE1,
-    SRTP_RCC_MODE2,
-    SRTP_RCC_MODE3
-}
-
-[CCode (cname = "int", cprefix = "SRTP_ENCR_", has_type_id = false)]
-public enum Encryption {
-    NULL,
-    AES_CM,
-    AES_F8
-}
-
-[CCode (cname = "int", cprefix = "SRTP_AUTH_", has_type_id = false)]
-public enum Authentication {
-    NULL,
-    HMAC_SHA1
-}
-
-[CCode (cname = "int", cprefix = "SRTP_PRF_", has_type_id = false)]
-public enum Prf {
-    AES_CM
-}
-
-}
\ No newline at end of file
diff --git a/plugins/crypto-vala/vapi/libsrtp2.vapi b/plugins/crypto-vala/vapi/libsrtp2.vapi
new file mode 100644
index 00000000..5ceedced
--- /dev/null
+++ b/plugins/crypto-vala/vapi/libsrtp2.vapi
@@ -0,0 +1,115 @@
+[CCode (cheader_filename = "srtp2/srtp.h")]
+namespace Srtp {
+public const uint MAX_TRAILER_LEN;
+
+public static ErrorStatus init();
+public static ErrorStatus shutdown();
+
+[Compact]
+[CCode (cname = "srtp_ctx_t", cprefix = "srtp_", free_function = "srtp_dealloc")]
+public class Context {
+    public static ErrorStatus create(out Context session, Policy? policy);
+
+    public ErrorStatus protect([CCode (type = "void*", array_length = false)] uint8[] rtp, ref int len);
+    public ErrorStatus unprotect([CCode (type = "void*", array_length = false)] uint8[] rtp, ref int len);
+
+    public ErrorStatus protect_rtcp([CCode (type = "void*", array_length = false)] uint8[] rtcp, ref int len);
+    public ErrorStatus unprotect_rtcp([CCode (type = "void*", array_length = false)] uint8[] rtcp, ref int len);
+
+    public ErrorStatus add_stream(ref Policy policy);
+    public ErrorStatus update_stream(ref Policy policy);
+    public ErrorStatus remove_stream(uint ssrc);
+    public ErrorStatus update(ref Policy policy);
+}
+
+[CCode (cname = "srtp_ssrc_t")]
+public struct Ssrc {
+    public SsrcType type;
+    public uint value;
+}
+
+[CCode (cname = "srtp_ssrc_type_t", cprefix = "ssrc_")]
+public enum SsrcType {
+    undefined, specific, any_inbound, any_outbound
+}
+
+[CCode (cname = "srtp_policy_t", destroy_function = "")]
+public struct Policy {
+    public Ssrc ssrc;
+    public CryptoPolicy rtp;
+    public CryptoPolicy rtcp;
+    [CCode (array_length = false)]
+    public uint8[] key;
+    public ulong num_master_keys;
+    public ulong window_size;
+    public int allow_repeat_tx;
+    [CCode (array_length_cname = "enc_xtn_hdr_count")]
+    public int[] enc_xtn_hdr;
+}
+
+[CCode (cname = "srtp_crypto_policy_t")]
+public struct CryptoPolicy {
+    public CipherType cipher_type;
+    public int cipher_key_len;
+    public AuthType auth_type;
+    public int auth_key_len;
+    public int auth_tag_len;
+    public SecurityServices sec_serv;
+
+    public void set_aes_cm_128_hmac_sha1_80();
+    public void set_aes_cm_128_hmac_sha1_32();
+    public void set_aes_cm_128_null_auth();
+    public void set_aes_cm_192_hmac_sha1_32();
+    public void set_aes_cm_192_hmac_sha1_80();
+    public void set_aes_cm_192_null_auth();
+    public void set_aes_cm_256_hmac_sha1_32();
+    public void set_aes_cm_256_hmac_sha1_80();
+    public void set_aes_cm_256_null_auth();
+    public void set_aes_gcm_128_16_auth();
+    public void set_aes_gcm_128_8_auth();
+    public void set_aes_gcm_128_8_only_auth();
+    public void set_aes_gcm_256_16_auth();
+    public void set_aes_gcm_256_8_auth();
+    public void set_aes_gcm_256_8_only_auth();
+    public void set_null_cipher_hmac_null();
+    public void set_null_cipher_hmac_sha1_80();
+
+    public void set_rtp_default();
+    public void set_rtcp_default();
+
+    public void set_from_profile_for_rtp(Profile profile);
+    public void set_from_profile_for_rtcp(Profile profile);
+}
+
+[CCode (cname = "srtp_profile_t", cprefix = "srtp_profile_")]
+public enum Profile {
+    reserved, aes128_cm_sha1_80, aes128_cm_sha1_32, null_sha1_80, null_sha1_32, aead_aes_128_gcm, aead_aes_256_gcm
+}
+
+[CCode (cname = "srtp_cipher_type_id_t")]
+public struct CipherType : uint32 {}
+
+[CCode (cname = "srtp_auth_type_id_t")]
+public struct AuthType : uint32 {}
+
+[CCode (cname = "srtp_sec_serv_t", cprefix = "sec_serv_")]
+public enum SecurityServices {
+    none, conf, auth, conf_and_auth;
+}
+
+[CCode (cname = "srtp_err_status_t", cprefix = "srtp_err_status_", has_type_id = false)]
+public enum ErrorStatus {
+    ok, fail, bad_param, alloc_fail, dealloc_fail, init_fail, terminus, auth_fail, cipher_fail, replay_fail, algo_fail, no_such_op, no_ctx, cant_check, key_expired, socket_err, signal_err, nonce_bad, encode_err, semaphore_err, pfkey_err, bad_mki, pkt_idx_old, pkt_idx_adv
+}
+
+[CCode (cname = "srtp_log_level_t", cprefix = "srtp_log_level_", has_type_id = false)]
+public enum LogLevel {
+    error, warning, info, debug
+}
+
+[CCode (cname = "srtp_log_handler_func_t")]
+public delegate void LogHandler(LogLevel level, string msg);
+
+public static ErrorStatus install_log_handler(LogHandler func);
+
+}
\ No newline at end of file
diff --git a/plugins/ice/CMakeLists.txt b/plugins/ice/CMakeLists.txt
index 38025aa0..392a202f 100644
--- a/plugins/ice/CMakeLists.txt
+++ b/plugins/ice/CMakeLists.txt
@@ -20,7 +20,7 @@ CUSTOM_VAPIS
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/dino.vapi
     ${CMAKE_BINARY_DIR}/exports/qlite.vapi
-    ${CMAKE_BINARY_DIR}/exports/crypto.vapi
+    ${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi
 PACKAGES
     ${ICE_PACKAGES}
 OPTIONS
diff --git a/plugins/ice/src/dtls_srtp.vala b/plugins/ice/src/dtls_srtp.vala
index a21c242b..b742ccab 100644
--- a/plugins/ice/src/dtls_srtp.vala
+++ b/plugins/ice/src/dtls_srtp.vala
@@ -12,8 +12,7 @@ public class DtlsSrtp {
     private uint pull_timeout = uint.MAX;
     private string peer_fingerprint;
 
-    private Crypto.Srtp.Session encrypt_session;
-    private Crypto.Srtp.Session decrypt_session;
+    private Crypto.Srtp.Session srtp_session = new Crypto.Srtp.Session();
 
     public static DtlsSrtp setup() throws GLib.Error {
         var obj = new DtlsSrtp();
@@ -30,9 +29,19 @@ public class DtlsSrtp {
     }
 
     public uint8[] process_incoming_data(uint component_id, uint8[] data) {
-        if (decrypt_session != null) {
-            if (component_id == 1) return decrypt_session.decrypt_rtp(data);
-            if (component_id == 2) return decrypt_session.decrypt_rtcp(data);
+        if (srtp_session.has_decrypt) {
+            try {
+                if (component_id == 1) {
+                    if (data.length >= 2 && data[1] >= 192 && data[1] < 224) {
+                        return srtp_session.decrypt_rtcp(data);
+                    }
+                    return srtp_session.decrypt_rtp(data);
+                }
+                if (component_id == 2) return srtp_session.decrypt_rtcp(data);
+            } catch (Error e) {
+                warning("%s (%d)", e.message, e.code);
+                return null;
+            }
         } else if (component_id == 1) {
             on_data_rec(data);
         }
@@ -40,9 +49,19 @@ public class DtlsSrtp {
     }
 
     public uint8[] process_outgoing_data(uint component_id, uint8[] data) {
-        if (encrypt_session != null) {
-            if (component_id == 1) return encrypt_session.encrypt_rtp(data);
-            if (component_id == 2) return encrypt_session.encrypt_rtcp(data);
+        if (srtp_session.has_encrypt) {
+            try {
+                if (component_id == 1) {
+                    if (data.length >= 2 && data[1] >= 192 && data[1] < 224) {
+                        return srtp_session.encrypt_rtcp(data);
+                    }
+                    return srtp_session.encrypt_rtp(data);
+                }
+                if (component_id == 2) return srtp_session.encrypt_rtcp(data);
+            } catch (Error e) {
+                warning("%s (%d)", e.message, e.code);
+                return null;
+            }
         }
         return null;
     }
@@ -123,19 +142,13 @@ public class DtlsSrtp {
             warning("SRTP client/server key/salt null");
         }
 
-        Crypto.Srtp.Session encrypt_session = new Crypto.Srtp.Session(Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, 10, Crypto.Srtp.Prf.AES_CM, 0);
-        Crypto.Srtp.Session decrypt_session = new Crypto.Srtp.Session(Crypto.Srtp.Encryption.AES_CM, Crypto.Srtp.Authentication.HMAC_SHA1, 10, Crypto.Srtp.Prf.AES_CM, 0);
-
         if (server) {
-            encrypt_session.setkey(server_key.extract(), server_salt.extract());
-            decrypt_session.setkey(client_key.extract(), client_salt.extract());
+            srtp_session.set_encryption_key(Crypto.Srtp.AES_CM_128_HMAC_SHA1_80, server_key.extract(), server_salt.extract());
+            srtp_session.set_decryption_key(Crypto.Srtp.AES_CM_128_HMAC_SHA1_80, client_key.extract(), client_salt.extract());
         } else {
-            encrypt_session.setkey(client_key.extract(), client_salt.extract());
-            decrypt_session.setkey(server_key.extract(), server_salt.extract());
+            srtp_session.set_encryption_key(Crypto.Srtp.AES_CM_128_HMAC_SHA1_80, client_key.extract(), client_salt.extract());
+            srtp_session.set_decryption_key(Crypto.Srtp.AES_CM_128_HMAC_SHA1_80, server_key.extract(), server_salt.extract());
         }
-
-        this.encrypt_session = (owned)encrypt_session;
-        this.decrypt_session = (owned)decrypt_session;
     }
 
     private static ssize_t pull_function(void* transport_ptr, uint8[] buffer) {
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 8ce2a7c6..c6888459 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -19,7 +19,7 @@ SOURCES
     src/video_widget.vala
     src/register_plugin.vala
 CUSTOM_VAPIS
-    ${CMAKE_BINARY_DIR}/exports/crypto.vapi
+    ${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/dino.vapi
     ${CMAKE_BINARY_DIR}/exports/qlite.vapi
diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala
index 77080a09..bedd6f8a 100644
--- a/plugins/rtp/src/stream.vala
+++ b/plugins/rtp/src/stream.vala
@@ -53,8 +53,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     private Gst.Pad send_rtp_sink_pad;
     private Gst.Pad send_rtp_src_pad;
 
-    private Crypto.Srtp.Session? local_crypto_session;
-    private Crypto.Srtp.Session? remote_crypto_session;
+    private Crypto.Srtp.Session? crypto_session = new Crypto.Srtp.Session();
 
     public Stream(Plugin plugin, Xmpp.Xep.Jingle.Content content) {
         base(content);
@@ -148,15 +147,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     }
 
     private void prepare_local_crypto() {
-        if (local_crypto != null && local_crypto_session == null) {
-            local_crypto_session = new Crypto.Srtp.Session(
-                    local_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? Crypto.Srtp.Encryption.AES_F8 : Crypto.Srtp.Encryption.AES_CM,
-                    Crypto.Srtp.Authentication.HMAC_SHA1,
-                    local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10,
-                    Crypto.Srtp.Prf.AES_CM,
-                    0
-            );
-            local_crypto_session.setkey(local_crypto.key, local_crypto.salt);
+        if (local_crypto != null && !crypto_session.has_encrypt) {
+            crypto_session.set_encryption_key(local_crypto.crypto_suite, local_crypto.key, local_crypto.salt);
             debug("Setting up encryption with key params %s", local_crypto.key_params);
         }
     }
@@ -172,15 +164,19 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         buffer.extract_dup(0, buffer.get_size(), out data);
         prepare_local_crypto();
         if (sink == send_rtp) {
-            if (local_crypto_session != null) {
-                data = local_crypto_session.encrypt_rtp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10);
+            if (crypto_session.has_encrypt) {
+                data = crypto_session.encrypt_rtp(data);
             }
             on_send_rtp_data(new Bytes.take(data));
         } else if (sink == send_rtcp) {
-            if (local_crypto_session != null) {
-                data = local_crypto_session.encrypt_rtcp(data, local_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10);
+            if (crypto_session.has_encrypt) {
+                data = crypto_session.encrypt_rtcp(data);
+            }
+            if (rtcp_mux) {
+                on_send_rtp_data(new Bytes.take(data));
+            } else {
+                on_send_rtcp_data(new Bytes.take(data));
             }
-            on_send_rtcp_data(new Bytes.take(data));
         } else {
             warning("unknown sample");
         }
@@ -283,25 +279,22 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     }
 
     private void prepare_remote_crypto() {
-        if (remote_crypto != null && remote_crypto_session == null) {
-            remote_crypto_session = new Crypto.Srtp.Session(
-                    remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.F8_128_HMAC_SHA1_80 ? Crypto.Srtp.Encryption.AES_F8 : Crypto.Srtp.Encryption.AES_CM,
-                    Crypto.Srtp.Authentication.HMAC_SHA1,
-                    remote_crypto.crypto_suite == Xep.JingleRtp.Crypto.AES_CM_128_HMAC_SHA1_32 ? 4 : 10,
-                    Crypto.Srtp.Prf.AES_CM,
-                    0
-            );
-            remote_crypto_session.setkey(remote_crypto.key, remote_crypto.salt);
+        if (remote_crypto != null && crypto_session.has_decrypt) {
+            crypto_session.set_decryption_key(remote_crypto.crypto_suite, remote_crypto.key, remote_crypto.salt);
             debug("Setting up decryption with key params %s", remote_crypto.key_params);
         }
     }
 
     public override void on_recv_rtp_data(Bytes bytes) {
+        if (rtcp_mux && bytes.length >= 2 && bytes.get(1) >= 192 && bytes.get(1) < 224) {
+            on_recv_rtcp_data(bytes);
+            return;
+        }
         prepare_remote_crypto();
         uint8[] data = bytes.get_data();
-        if (remote_crypto_session != null) {
+        if (crypto_session.has_decrypt) {
             try {
-                data = remote_crypto_session.decrypt_rtp(data);
+                data = crypto_session.decrypt_rtp(data);
             } catch (Error e) {
                 warning("%s (%d)", e.message, e.code);
             }
@@ -314,9 +307,9 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     public override void on_recv_rtcp_data(Bytes bytes) {
         prepare_remote_crypto();
         uint8[] data = bytes.get_data();
-        if (remote_crypto_session != null) {
+        if (crypto_session.has_decrypt) {
             try {
-                data = remote_crypto_session.decrypt_rtcp(data);
+                data = crypto_session.decrypt_rtcp(data);
             } catch (Error e) {
                 warning("%s (%d)", e.message, e.code);
             }
-- 
cgit v1.2.3-70-g09d2


From 3880628de4785db4c0a03a79a0c486507fe9b1a8 Mon Sep 17 00:00:00 2001
From: Marvin W <git@larma.de>
Date: Thu, 29 Apr 2021 15:46:06 +0200
Subject: Video optimizations

---
 cmake/FindGstRtp.cmake                             |  14 +
 plugins/rtp/CMakeLists.txt                         |   4 +-
 plugins/rtp/src/codec_util.vala                    | 115 +++-
 plugins/rtp/src/device.vala                        |   9 +-
 plugins/rtp/src/module.vala                        | 133 +++--
 plugins/rtp/src/plugin.vala                        |  14 +
 plugins/rtp/src/stream.vala                        | 220 +++++++-
 plugins/rtp/src/video_widget.vala                  |  10 +-
 plugins/rtp/vapi/gstreamer-rtp-1.0.vapi            | 625 +++++++++++++++++++++
 .../xep/0167_jingle_rtp/content_parameters.vala    |  40 +-
 .../xep/0167_jingle_rtp/jingle_rtp_module.vala     |   5 +
 .../module/xep/0167_jingle_rtp/payload_type.vala   |  49 +-
 .../src/module/xep/0167_jingle_rtp/stream.vala     |   7 +
 13 files changed, 1126 insertions(+), 119 deletions(-)
 create mode 100644 cmake/FindGstRtp.cmake
 create mode 100644 plugins/rtp/vapi/gstreamer-rtp-1.0.vapi

(limited to 'cmake')

diff --git a/cmake/FindGstRtp.cmake b/cmake/FindGstRtp.cmake
new file mode 100644
index 00000000..0756a985
--- /dev/null
+++ b/cmake/FindGstRtp.cmake
@@ -0,0 +1,14 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GstRtp
+    PKG_CONFIG_NAME gstreamer-rtp-1.0
+    LIB_NAMES gstrtp
+    LIB_DIR_HINTS gstreamer-1.0
+    INCLUDE_NAMES gst/rtp/rtp.h
+    INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include gstreamer-rtp-1.0 gstreamer-rtp-1.0/include
+    DEPENDS Gst
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstRtp
+    REQUIRED_VARS GstRtp_LIBRARY
+    VERSION_VAR GstRtp_VERSION)
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 76d6e66d..92ec1b97 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -1,3 +1,4 @@
+find_package(GstRtp REQUIRED)
 find_packages(RTP_PACKAGES REQUIRED
     Gee
     GLib
@@ -27,6 +28,7 @@ CUSTOM_VAPIS
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/dino.vapi
     ${CMAKE_BINARY_DIR}/exports/qlite.vapi
+    ${CMAKE_CURRENT_SOURCE_DIR}/vapi/gstreamer-rtp-1.0.vapi
 PACKAGES
     ${RTP_PACKAGES}
 DEFINITIONS
@@ -35,7 +37,7 @@ DEFINITIONS
 
 add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
 add_library(rtp SHARED ${RTP_VALA_C})
-target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES})
+target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0)
 set_target_properties(rtp PROPERTIES PREFIX "")
 set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
 
diff --git a/plugins/rtp/src/codec_util.vala b/plugins/rtp/src/codec_util.vala
index 6bd465c1..7537c11d 100644
--- a/plugins/rtp/src/codec_util.vala
+++ b/plugins/rtp/src/codec_util.vala
@@ -6,7 +6,7 @@ public class Dino.Plugins.Rtp.CodecUtil {
     private Set<string> supported_elements = new HashSet<string>();
     private Set<string> unsupported_elements = new HashSet<string>();
 
-    public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type) {
+    public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type, bool incoming) {
         Gst.Caps caps = new Gst.Caps.simple("application/x-rtp",
                 "media", typeof(string), media,
                 "payload", typeof(int), payload_type.id);
@@ -19,6 +19,15 @@ public class Dino.Plugins.Rtp.CodecUtil {
         if (payload_type.name != null) {
             s.set("encoding-name", typeof(string), payload_type.name.up());
         }
+        if (incoming) {
+            foreach (JingleRtp.RtcpFeedback rtcp_fb in payload_type.rtcp_fbs) {
+                if (rtcp_fb.subtype == null) {
+                    s.set(@"rtcp-fb-$(rtcp_fb.type_)", typeof(bool), true);
+                } else {
+                    s.set(@"rtcp-fb-$(rtcp_fb.type_)-$(rtcp_fb.subtype)", typeof(bool), true);
+                }
+            }
+        }
         return caps;
     }
 
@@ -122,32 +131,82 @@ public class Dino.Plugins.Rtp.CodecUtil {
         return new string[0];
     }
 
-    public static string? get_encode_prefix(string media, string codec, string encode) {
+    public static string? get_encode_prefix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
         if (encode == "msdkh264enc") return "video/x-raw,format=NV12 ! ";
         if (encode == "vaapih264enc") return "video/x-raw,format=NV12 ! ";
         return null;
     }
 
-    public static string? get_encode_suffix(string media, string codec, string encode) {
+    public static string? get_encode_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
         // H264
-        const string h264_suffix = " ! video/x-h264,profile=constrained-baseline ! h264parse";
-        if (encode == "msdkh264enc") return @" bitrate=256 rate-control=vbr target-usage=7$h264_suffix";
-        if (encode == "vaapih264enc") return @" bitrate=256 quality-level=7 tune=low-power$h264_suffix";
-        if (encode == "x264enc") return @" byte-stream=1 bitrate=256 profile=baseline speed-preset=ultrafast tune=zerolatency$h264_suffix";
-        if (media == "video" && codec == "h264") return h264_suffix;
+        if (encode == "msdkh264enc") return @" rate-control=vbr";
+        if (encode == "vaapih264enc") return @" tune=low-power";
+        if (encode == "x264enc") return @" byte-stream=1 profile=baseline speed-preset=ultrafast tune=zerolatency";
 
         // VP8
-        if (encode == "msdkvp8enc") return " bitrate=256 rate-control=vbr target-usage=7";
-        if (encode == "vaapivp8enc") return " bitrate=256 rate-control=vbr quality-level=7";
-        if (encode == "vp8enc") return " target-bitrate=256000 deadline=1 error-resilient=1";
+        if (encode == "msdkvp8enc") return " rate-control=vbr";
+        if (encode == "vaapivp8enc") return " rate-control=vbr";
+        if (encode == "vp8enc") return " deadline=1 error-resilient=1";
 
         // OPUS
-        if (encode == "opusenc") return " audio-type=voice";
+        if (encode == "opusenc") {
+            if (payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " audio-type=voice inband-fec=true";
+            return " audio-type=voice";
+        }
 
         return null;
     }
 
-    public static string? get_decode_prefix(string media, string codec, string decode) {
+    public static string? get_encode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        // H264
+        if (media == "video" && codec == "h264") return " ! video/x-h264,profile=constrained-baseline ! h264parse";
+        return null;
+    }
+
+    public uint update_bitrate(string media, JingleRtp.PayloadType payload_type, Gst.Element encode_element, uint bitrate) {
+        Gst.Bin? encode_bin = encode_element as Gst.Bin;
+        if (encode_bin == null) return 0;
+        string? codec = get_codec_from_payload(media, payload_type);
+        string? encode_name = get_encode_element_name(media, codec);
+        if (encode_name == null) return 0;
+        Gst.Element encode = encode_bin.get_by_name(@"$(encode_bin.name)_encode");
+
+        bitrate = uint.min(2048000, bitrate);
+
+        switch (encode_name) {
+            case "msdkh264enc":
+            case "vaapih264enc":
+            case "x264enc":
+            case "msdkvp8enc":
+            case "vaapivp8enc":
+                bitrate = uint.min(2048000, bitrate);
+                encode.set("bitrate", bitrate);
+                return bitrate;
+            case "vp8enc":
+                bitrate = uint.min(2147483, bitrate);
+                encode.set("target-bitrate", bitrate * 1000);
+                return bitrate;
+        }
+
+        return 0;
+    }
+
+    public static string? get_decode_prefix(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) {
+        return null;
+    }
+
+    public static string? get_decode_args(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) {
+        if (decode == "opusdec" && payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " use-inband-fec=true";
+        if (decode == "vaapivp9dec" || decode == "vaapivp8dec" || decode == "vaapih264dec") return " max-errors=100";
+        return null;
+    }
+
+    public static string? get_decode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        return null;
+    }
+
+    public static string? get_depay_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        if (codec == "vp8") return " wait-for-keyframe=true";
         return null;
     }
 
@@ -195,21 +254,24 @@ public class Dino.Plugins.Rtp.CodecUtil {
         unsupported_elements.add(element_name);
     }
 
-    public string? get_decode_bin_description(string media, string? codec, string? element_name = null, string? name = null) {
+    public string? get_decode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) {
         if (codec == null) return null;
         string base_name = name ?? @"encode-$codec-$(Random.next_int())";
         string depay = get_depay_element_name(media, codec);
         string decode = element_name ?? get_decode_element_name(media, codec);
         if (depay == null || decode == null) return null;
-        string decode_prefix = get_decode_prefix(media, codec, decode) ?? "";
-        string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
-        return @"$depay name=$base_name-rtp-depay ! $decode_prefix$decode name=$base_name-decode ! $(media)convert name=$base_name-convert$resample";
+        string decode_prefix = get_decode_prefix(media, codec, decode, payload_type) ?? "";
+        string decode_args = get_decode_args(media, codec, decode, payload_type) ?? "";
+        string decode_suffix = get_decode_suffix(media, codec, decode, payload_type) ?? "";
+        string depay_args = get_depay_args(media, codec, decode, payload_type) ?? "";
+        string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : "";
+        return @"$depay$depay_args name=$(base_name)_rtp_depay ! $decode_prefix$decode$decode_args name=$(base_name)_$(codec)_decode$decode_suffix ! $(media)convert name=$(base_name)_convert$resample";
     }
 
     public Gst.Element? get_decode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) {
         string? codec = get_codec_from_payload(media, payload_type);
         string base_name = name ?? @"encode-$codec-$(Random.next_int())";
-        string? desc = get_decode_bin_description(media, codec, null, base_name);
+        string? desc = get_decode_bin_description(media, codec, payload_type, null, base_name);
         if (desc == null) return null;
         debug("Pipeline to decode %s %s: %s", media, codec, desc);
         Gst.Element bin = Gst.parse_bin_from_description(desc, true);
@@ -217,22 +279,23 @@ public class Dino.Plugins.Rtp.CodecUtil {
         return bin;
     }
 
-    public string? get_encode_bin_description(string media, string? codec, string? element_name = null, uint pt = 96, string? name = null) {
+    public string? get_encode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) {
         if (codec == null) return null;
-        string base_name = name ?? @"encode-$codec-$(Random.next_int())";
+        string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
         string pay = get_pay_element_name(media, codec);
         string encode = element_name ?? get_encode_element_name(media, codec);
         if (pay == null || encode == null) return null;
-        string encode_prefix = get_encode_prefix(media, codec, encode) ?? "";
-        string encode_suffix = get_encode_suffix(media, codec, encode) ?? "";
-        string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
-        return @"$(media)convert name=$base_name-convert$resample ! $encode_prefix$encode$encode_suffix ! $pay pt=$pt name=$base_name-rtp-pay";
+        string encode_prefix = get_encode_prefix(media, codec, encode, payload_type) ?? "";
+        string encode_args = get_encode_args(media, codec, encode, payload_type) ?? "";
+        string encode_suffix = get_encode_suffix(media, codec, encode, payload_type) ?? "";
+        string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : "";
+        return @"$(media)convert name=$(base_name)_convert$resample ! $encode_prefix$encode$encode_args name=$(base_name)_encode$encode_suffix ! $pay pt=$(payload_type != null ? payload_type.id : 96) name=$(base_name)_rtp_pay";
     }
 
     public Gst.Element? get_encode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) {
         string? codec = get_codec_from_payload(media, payload_type);
-        string base_name = name ?? @"encode-$codec-$(Random.next_int())";
-        string? desc = get_encode_bin_description(media, codec, null, payload_type.id, base_name);
+        string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
+        string? desc = get_encode_bin_description(media, codec, payload_type, null, base_name);
         if (desc == null) return null;
         debug("Pipeline to encode %s %s: %s", media, codec, desc);
         Gst.Element bin = Gst.parse_bin_from_description(desc, true);
diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala
index 3c9a38d2..785f853a 100644
--- a/plugins/rtp/src/device.vala
+++ b/plugins/rtp/src/device.vala
@@ -126,19 +126,20 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
         element = device.create_element(id);
         pipe.add(element);
         if (is_source) {
-            filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+            element.@set("do-timestamp", true);
+            filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             element.link(filter);
             if (media == "audio" && plugin.echoprobe != null) {
-                dsp = Gst.ElementFactory.make("webrtcdsp", @"$id-dsp");
+                dsp = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
                 if (dsp != null) {
                     dsp.@set("probe", plugin.echoprobe.name);
                     pipe.add(dsp);
                     filter.link(dsp);
                 }
             }
-            tee = Gst.ElementFactory.make("tee", @"$id-tee");
+            tee = Gst.ElementFactory.make("tee", @"tee_$id");
             tee.@set("allow-not-linked", true);
             pipe.add(tee);
             (dsp ?? filter).link(tee);
@@ -148,7 +149,7 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             element.@set("sync", false);
         }
         if (is_sink && media == "audio") {
-            filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+            filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             if (plugin.echoprobe != null) {
diff --git a/plugins/rtp/src/module.vala b/plugins/rtp/src/module.vala
index 231a9dde..52cc1880 100644
--- a/plugins/rtp/src/module.vala
+++ b/plugins/rtp/src/module.vala
@@ -63,7 +63,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return supported;
     }
 
-    private async bool supports(string media, JingleRtp.PayloadType payload_type) {
+    private async bool is_payload_supported(string media, JingleRtp.PayloadType payload_type) {
         string codec = CodecUtil.get_codec_from_payload(media, payload_type);
         if (codec == null) return false;
         if (unsupported_codecs.contains(codec)) return false;
@@ -77,7 +77,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
             return false;
         }
 
-        string encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+        string encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
         while (!(yield pipeline_works(media, encode_bin))) {
             debug("%s not suited for encoding %s", encode_element, codec);
             codec_util.mark_element_unsupported(encode_element);
@@ -87,11 +87,11 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
                 unsupported_codecs.add(codec);
                 return false;
             }
-            encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+            encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
         }
         debug("using %s to encode %s", encode_element, codec);
 
-        string decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+        string decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
         while (!(yield pipeline_works(media, @"$encode_bin ! $decode_bin"))) {
             debug("%s not suited for decoding %s", decode_element, codec);
             codec_util.mark_element_unsupported(decode_element);
@@ -101,7 +101,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
                 unsupported_codecs.add(codec);
                 return false;
             }
-            decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+            decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
         }
         debug("using %s to decode %s", decode_element, codec);
 
@@ -109,8 +109,21 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return true;
     }
 
+    public override bool is_header_extension_supported(string media, JingleRtp.HeaderExtension ext) {
+        if (media == "video" && ext.uri == "urn:3gpp:video-orientation") return true;
+        return false;
+    }
+
+    public override Gee.List<JingleRtp.HeaderExtension> get_suggested_header_extensions(string media) {
+        Gee.List<JingleRtp.HeaderExtension> exts = new ArrayList<JingleRtp.HeaderExtension>();
+        if (media == "video") {
+            exts.add(new JingleRtp.HeaderExtension(1, "urn:3gpp:video-orientation"));
+        }
+        return exts;
+    }
+
     public async void add_if_supported(Gee.List<JingleRtp.PayloadType> list, string media, JingleRtp.PayloadType payload_type) {
-        if (yield supports(media, payload_type)) {
+        if (yield is_payload_supported(media, payload_type)) {
             list.add(payload_type);
         }
     }
@@ -118,58 +131,34 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
     public override async Gee.List<JingleRtp.PayloadType> get_supported_payloads(string media) {
         Gee.List<JingleRtp.PayloadType> list = new ArrayList<JingleRtp.PayloadType>(JingleRtp.PayloadType.equals_func);
         if (media == "audio") {
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 2,
-                clockrate = 48000,
-                name = "opus",
-                id = 99
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 32000,
-                name = "speex",
-                id = 100
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 16000,
-                name = "speex",
-                id = 101
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "speex",
-                id = 102
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "PCMU",
-                id = 0
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "PCMA",
-                id = 8
-            });
+            var opus = new JingleRtp.PayloadType() { channels = 2, clockrate = 48000, name = "opus", id = 99 };
+            opus.parameters["useinbandfec"] = "1";
+            var speex32 = new JingleRtp.PayloadType() { channels = 1, clockrate = 32000, name = "speex", id = 100 };
+            var speex16 = new JingleRtp.PayloadType() { channels = 1, clockrate = 16000, name = "speex", id = 101 };
+            var speex8 = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "speex", id = 102 };
+            var pcmu = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMU", id = 0 };
+            var pcma = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMA", id = 8 };
+            yield add_if_supported(list, media, opus);
+            yield add_if_supported(list, media, speex32);
+            yield add_if_supported(list, media, speex16);
+            yield add_if_supported(list, media, speex8);
+            yield add_if_supported(list, media, pcmu);
+            yield add_if_supported(list, media, pcma);
         } else if (media == "video") {
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "H264",
-                id = 96
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "VP9",
-                id = 97
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "VP8",
-                id = 98
-            });
+            var h264 = new JingleRtp.PayloadType() { clockrate = 90000, name = "H264", id = 96 };
+            var vp9 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP9", id = 97 };
+            var vp8 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP8", id = 98 };
+            var rtcp_fbs = new ArrayList<JingleRtp.RtcpFeedback>();
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("goog-remb"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("ccm", "fir"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack", "pli"));
+            h264.rtcp_fbs.add_all(rtcp_fbs);
+            vp9.rtcp_fbs.add_all(rtcp_fbs);
+            vp8.rtcp_fbs.add_all(rtcp_fbs);
+            yield add_if_supported(list, media, h264);
+            yield add_if_supported(list, media, vp9);
+            yield add_if_supported(list, media, vp8);
         } else {
             warning("Unsupported media type: %s", media);
         }
@@ -179,11 +168,15 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
     public override async JingleRtp.PayloadType? pick_payload_type(string media, Gee.List<JingleRtp.PayloadType> payloads) {
         if (media == "audio") {
             foreach (JingleRtp.PayloadType type in payloads) {
-                if (yield supports(media, type)) return type;
+                if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone());
             }
         } else if (media == "video") {
+            // We prefer H.264 (best support for hardware acceleration and good overall codec quality)
+            JingleRtp.PayloadType? h264 = payloads.first_match((it) => it.name.up() == "H264");
+            if (h264 != null && yield is_payload_supported(media, h264)) return adjust_payload_type(media, h264.clone());
+            // Take first of the list that we do support otherwise
             foreach (JingleRtp.PayloadType type in payloads) {
-                if (yield supports(media, type)) return type;
+                if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone());
             }
         } else {
             warning("Unsupported media type: %s", media);
@@ -191,6 +184,28 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return null;
     }
 
+    public JingleRtp.PayloadType adjust_payload_type(string media, JingleRtp.PayloadType type) {
+        var iter = type.rtcp_fbs.iterator();
+        while (iter.next()) {
+            var fb = iter.@get();
+            switch (fb.type_) {
+                case "goog-remb":
+                    if (fb.subtype != null) iter.remove();
+                    break;
+                case "ccm":
+                    if (fb.subtype != "fir") iter.remove();
+                    break;
+                case "nack":
+                    if (fb.subtype != null && fb.subtype != "pli") iter.remove();
+                    break;
+                default:
+                    iter.remove();
+                    break;
+            }
+        }
+        return type;
+    }
+
     public override JingleRtp.Stream create_stream(Jingle.Content content) {
         return plugin.open_stream(content);
     }
diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala
index 40ad1e0f..f0ad7db2 100644
--- a/plugins/rtp/src/plugin.vala
+++ b/plugins/rtp/src/plugin.vala
@@ -65,6 +65,9 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
         }
         rtpbin.pad_added.connect(on_rtp_pad_added);
         rtpbin.@set("latency", 100);
+        rtpbin.@set("do-lost", true);
+        rtpbin.@set("do-sync-event", true);
+        rtpbin.@set("drop-on-latency", true);
         rtpbin.connect("signal::request-pt-map", request_pt_map, this);
         pipe.add(rtpbin);
 
@@ -160,6 +163,17 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
             case Gst.MessageType.QOS:
                 // Ignore
                 break;
+            case Gst.MessageType.LATENCY:
+                if (message.src != null && message.src.name != null && message.src is Gst.Element) {
+                    Gst.Query latency_query = new Gst.Query.latency();
+                    if (((Gst.Element)message.src).query(latency_query)) {
+                        bool live;
+                        Gst.ClockTime min_latency, max_latency;
+                        latency_query.parse_latency(out live, out min_latency, out max_latency);
+                        debug("Latency message from %s: live=%s, min_latency=%s, max_latency=%s", message.src.name, live.to_string(), min_latency.to_string(), max_latency.to_string());
+                    }
+                }
+                break;
             default:
                 debug("Pipe bus message: %s", message.type.to_string());
                 break;
diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala
index 3a63f3fa..23634aa3 100644
--- a/plugins/rtp/src/stream.vala
+++ b/plugins/rtp/src/stream.vala
@@ -19,9 +19,12 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     private Gst.App.Src recv_rtp;
     private Gst.App.Src recv_rtcp;
     private Gst.Element encode;
+    private Gst.RTP.BasePayload encode_pay;
     private Gst.Element decode;
+    private Gst.RTP.BaseDepayload decode_depay;
     private Gst.Element input;
     private Gst.Element output;
+    private Gst.Element session;
 
     private Device _input_device;
     public Device input_device { get { return _input_device; } set {
@@ -85,15 +88,15 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
 
         // Create app elements
-        send_rtp = Gst.ElementFactory.make("appsink", @"rtp-sink-$rtpid") as Gst.App.Sink;
+        send_rtp = Gst.ElementFactory.make("appsink", @"rtp_sink_$rtpid") as Gst.App.Sink;
         send_rtp.async = false;
-        send_rtp.caps = CodecUtil.get_caps(media, payload_type);
+        send_rtp.caps = CodecUtil.get_caps(media, payload_type, false);
         send_rtp.emit_signals = true;
         send_rtp.sync = false;
         send_rtp.new_sample.connect(on_new_sample);
         pipe.add(send_rtp);
 
-        send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp-sink-$rtpid") as Gst.App.Sink;
+        send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp_sink_$rtpid") as Gst.App.Sink;
         send_rtcp.async = false;
         send_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
         send_rtcp.emit_signals = true;
@@ -101,14 +104,14 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         send_rtcp.new_sample.connect(on_new_sample);
         pipe.add(send_rtcp);
 
-        recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp-src-$rtpid") as Gst.App.Src;
-        recv_rtp.caps = CodecUtil.get_caps(media, payload_type);
+        recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp_src_$rtpid") as Gst.App.Src;
+        recv_rtp.caps = CodecUtil.get_caps(media, payload_type, true);
         recv_rtp.do_timestamp = true;
         recv_rtp.format = Gst.Format.TIME;
         recv_rtp.is_live = true;
         pipe.add(recv_rtp);
 
-        recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp-src-$rtpid") as Gst.App.Src;
+        recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp_src_$rtpid") as Gst.App.Src;
         recv_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
         recv_rtcp.do_timestamp = true;
         recv_rtcp.format = Gst.Format.TIME;
@@ -122,7 +125,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         recv_rtcp.get_static_pad("src").link(recv_rtcp_sink_pad);
 
         // Connect input
-        encode = codec_util.get_encode_bin(media, payload_type, @"encode-$rtpid");
+        encode = codec_util.get_encode_bin(media, payload_type, @"encode_$rtpid");
+        encode_pay = (Gst.RTP.BasePayload)((Gst.Bin)encode).get_by_name(@"encode_$(rtpid)_rtp_pay");
         pipe.add(encode);
         send_rtp_sink_pad = rtpbin.get_request_pad(@"send_rtp_sink_$rtpid");
         encode.get_static_pad("src").link(send_rtp_sink_pad);
@@ -131,7 +135,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
 
         // Connect output
-        decode = codec_util.get_decode_bin(media, payload_type, @"decode-$rtpid");
+        decode = codec_util.get_decode_bin(media, payload_type, @"decode_$rtpid");
+        decode_depay = (Gst.RTP.BaseDepayload)((Gst.Bin)encode).get_by_name(@"decode_$(rtpid)_rtp_depay");
         pipe.add(decode);
         if (output != null) {
             decode.link(output);
@@ -144,6 +149,110 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         created = true;
         push_recv_data = true;
         plugin.unpause();
+
+        GLib.Signal.emit_by_name(rtpbin, "get-session", rtpid, out session);
+        if (session != null && payload_type.rtcp_fbs.any_match((it) => it.type_ == "goog-remb")) {
+            Object internal_session;
+            session.@get("internal-session", out internal_session);
+            if (internal_session != null) {
+                internal_session.connect("signal::on-feedback-rtcp", on_feedback_rtcp, this);
+            }
+            Timeout.add(1000, () => remb_adjust());
+        }
+        if (media == "video") {
+            codec_util.update_bitrate(media, payload_type, encode, 256);
+        }
+    }
+
+    private uint remb = 256;
+    private int last_packets_lost = -1;
+    private uint64 last_packets_received;
+    private uint64 last_octets_received;
+    private bool remb_adjust() {
+        unowned Gst.Structure? stats;
+        if (session == null) {
+            debug("Session for %u finished, turning off remb adjustment", rtpid);
+            return Source.REMOVE;
+        }
+        session.get("stats", out stats);
+        if (stats == null) {
+            warning("No stats for session %u", rtpid);
+            return Source.REMOVE;
+        }
+        unowned ValueArray? source_stats;
+        stats.get("source-stats", typeof(ValueArray), out source_stats);
+        if (source_stats == null) {
+            warning("No source-stats for session %u", rtpid);
+            return Source.REMOVE;
+        }
+        foreach (Value value in source_stats.values) {
+            unowned Gst.Structure source_stat = (Gst.Structure) value.get_boxed();
+            uint ssrc;
+            if (!source_stat.get_uint("ssrc", out ssrc)) continue;
+            if (ssrc.to_string() == participant_ssrc) {
+                int packets_lost;
+                uint64 packets_received, octets_received;
+                source_stat.get_int("packets-lost", out packets_lost);
+                source_stat.get_uint64("packets-received", out packets_received);
+                source_stat.get_uint64("octets-received", out octets_received);
+                int new_lost = packets_lost - last_packets_lost;
+                uint64 new_received = packets_received - last_packets_received;
+                uint64 new_octets = octets_received - last_octets_received;
+                if (new_received == 0) continue;
+                last_packets_lost = packets_lost;
+                last_packets_received = packets_received;
+                last_octets_received = octets_received;
+                double loss_rate = (double)new_lost / (double)(new_lost + new_received);
+                if (new_lost <= 0 || loss_rate < 0.02) {
+                    remb = (uint)(1.08 * (double)remb);
+                } else if (loss_rate > 0.1) {
+                    remb = (uint)((1.0 - 0.5 * loss_rate) * (double)remb);
+                }
+                remb = uint.max(remb, (uint)((new_octets * 8) / 1000));
+                remb = uint.max(16, remb); // Never go below 16
+                uint8[] data = new uint8[] {
+                    143, 206, 0, 5,
+                    0, 0, 0, 0,
+                    0, 0, 0, 0,
+                    'R', 'E', 'M', 'B',
+                    1, 0, 0, 0,
+                    0, 0, 0, 0
+                };
+                data[4] = (uint8)((encode_pay.ssrc >> 24) & 0xff);
+                data[5] = (uint8)((encode_pay.ssrc >> 16) & 0xff);
+                data[6] = (uint8)((encode_pay.ssrc >> 8) & 0xff);
+                data[7] = (uint8)(encode_pay.ssrc & 0xff);
+                uint8 br_exp = 0;
+                uint32 br_mant = remb * 1000;
+                uint8 bits = (uint8)Math.log2(br_mant);
+                if (bits > 16) {
+                    br_exp = (uint8)bits - 16;
+                    br_mant = br_mant >> br_exp;
+                }
+                data[17] = (uint8)((br_exp << 2) | ((br_mant >> 16) & 0x3));
+                data[18] = (uint8)((br_mant >> 8) & 0xff);
+                data[19] = (uint8)(br_mant & 0xff);
+                data[20] = (uint8)((ssrc >> 24) & 0xff);
+                data[21] = (uint8)((ssrc >> 16) & 0xff);
+                data[22] = (uint8)((ssrc >> 8) & 0xff);
+                data[23] = (uint8)(ssrc & 0xff);
+                encrypt_and_send_rtcp(data);
+            }
+        }
+        return Source.CONTINUE;
+    }
+
+    private static void on_feedback_rtcp(Gst.Element session, uint type, uint fbtype, uint sender_ssrc, uint media_ssrc, Gst.Buffer? fci, Stream self) {
+        if (type == 206 && fbtype == 15 && fci != null && sender_ssrc.to_string() == self.participant_ssrc) {
+            // https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
+            uint8[] data;
+            fci.extract_dup(0, fci.get_size(), out data);
+            if (data[0] != 'R' || data[1] != 'E' || data[2] != 'M' || data[3] != 'B') return;
+            uint8 br_exp = data[5] >> 2;
+            uint32 br_mant = (((uint32)data[5] & 0x3) << 16) + ((uint32)data[6] << 8) + (uint32)data[7];
+            uint bitrate = (br_mant << br_exp) / 1000;
+            self.codec_util.update_bitrate(self.media, self.payload_type, self.encode, bitrate * 8);
+        }
     }
 
     private void prepare_local_crypto() {
@@ -167,22 +276,26 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
             if (crypto_session.has_encrypt) {
                 data = crypto_session.encrypt_rtp(data);
             }
-            on_send_rtp_data(new Bytes.take(data));
+            on_send_rtp_data(new Bytes.take((owned) data));
         } else if (sink == send_rtcp) {
-            if (crypto_session.has_encrypt) {
-                data = crypto_session.encrypt_rtcp(data);
-            }
-            if (rtcp_mux) {
-                on_send_rtp_data(new Bytes.take(data));
-            } else {
-                on_send_rtcp_data(new Bytes.take(data));
-            }
+            encrypt_and_send_rtcp((owned) data);
         } else {
             warning("unknown sample");
         }
         return Gst.FlowReturn.OK;
     }
 
+    private void encrypt_and_send_rtcp(owned uint8[] data) {
+        if (crypto_session.has_encrypt) {
+            data = crypto_session.encrypt_rtcp(data);
+        }
+        if (rtcp_mux) {
+            on_send_rtp_data(new Bytes.take((owned) data));
+        } else {
+            on_send_rtcp_data(new Bytes.take((owned) data));
+        }
+    }
+
     private static Gst.PadProbeReturn drop_probe() {
         return Gst.PadProbeReturn.DROP;
     }
@@ -211,6 +324,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         encode.get_static_pad("src").unlink(send_rtp_sink_pad);
         pipe.remove(encode);
         encode = null;
+        encode_pay = null;
 
         // Disconnect RTP sending
         if (send_rtp_src_pad != null) {
@@ -243,6 +357,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         decode.set_state(Gst.State.NULL);
         pipe.remove(decode);
         decode = null;
+        decode_depay = null;
         output = null;
 
         // Disconnect output device
@@ -276,6 +391,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         send_rtcp_src_pad = null;
         send_rtp_src_pad = null;
         recv_rtp_src_pad = null;
+
+        session = null;
     }
 
     private void prepare_remote_crypto() {
@@ -285,6 +402,9 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
     }
 
+    private uint16 previous_video_orientation_degree = uint16.MAX;
+    public signal void video_orientation_changed(uint16 degree);
+
     public override void on_recv_rtp_data(Bytes bytes) {
         if (rtcp_mux && bytes.length >= 2 && bytes.get(1) >= 192 && bytes.get(1) < 224) {
             on_recv_rtcp_data(bytes);
@@ -301,6 +421,33 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
         if (push_recv_data) {
             Gst.Buffer buffer = new Gst.Buffer.wrapped((owned) data);
+            Gst.RTP.Buffer rtp_buffer;
+            if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) {
+                if (rtp_buffer.get_extension()) {
+                    Xmpp.Xep.JingleRtp.HeaderExtension? ext = header_extensions.first_match((it) => it.uri == "urn:3gpp:video-orientation");
+                    if (ext != null) {
+                        unowned uint8[] extension_data;
+                        if (rtp_buffer.get_extension_onebyte_header(ext.id, 0, out extension_data) && extension_data.length == 1) {
+                            bool camera = (extension_data[0] & 0x8) > 0;
+                            bool flip = (extension_data[0] & 0x4) > 0;
+                            uint8 rotation = extension_data[0] & 0x3;
+                            uint16 rotation_degree = uint16.MAX;
+                            switch(rotation) {
+                                case 0: rotation_degree = 0; break;
+                                case 1: rotation_degree = 90; break;
+                                case 2: rotation_degree = 180; break;
+                                case 3: rotation_degree = 270; break;
+                            }
+                            if (rotation_degree != previous_video_orientation_degree) {
+                                video_orientation_changed(rotation_degree);
+                                previous_video_orientation_degree = rotation_degree;
+                            }
+                        }
+                    }
+                }
+                rtp_buffer.unmap();
+            }
+
             // FIXME: VAPI file in Vala < 0.49.1 has a bug that results in broken ownership of buffer in push_buffer()
             // We workaround by using the plain signal. The signal unfortunately will cause an unnecessary copy of
             // the underlying buffer, so and some point we should move over to the new version (once we require
@@ -449,6 +596,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
 public class Dino.Plugins.Rtp.VideoStream : Stream {
     private Gee.List<Gst.Element> outputs = new ArrayList<Gst.Element>();
     private Gst.Element output_tee;
+    private Gst.Element rotate;
+    private ulong video_orientation_changed_handler;
 
     public VideoStream(Plugin plugin, Xmpp.Xep.Jingle.Content content) {
         base(plugin, content);
@@ -456,11 +605,15 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
     }
 
     public override void create() {
+        video_orientation_changed_handler = video_orientation_changed.connect(on_video_orientation_changed);
         plugin.pause();
-        output_tee = Gst.ElementFactory.make("tee", null);
+        rotate = Gst.ElementFactory.make("videoflip", @"video_rotate_$rtpid");
+        pipe.add(rotate);
+        output_tee = Gst.ElementFactory.make("tee", @"video_tee_$rtpid");
         output_tee.@set("allow-not-linked", true);
         pipe.add(output_tee);
-        add_output(output_tee);
+        rotate.link(output_tee);
+        add_output(rotate);
         base.create();
         foreach (Gst.Element output in outputs) {
             output_tee.link(output);
@@ -468,19 +621,44 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
         plugin.unpause();
     }
 
+    private void on_video_orientation_changed(uint16 degree) {
+        if (rotate != null) {
+            switch (degree) {
+                case 0:
+                    rotate.@set("method", 0);
+                    break;
+                case 90:
+                    rotate.@set("method", 1);
+                    break;
+                case 180:
+                    rotate.@set("method", 2);
+                    break;
+                case 270:
+                    rotate.@set("method", 3);
+                    break;
+            }
+        }
+    }
+
     public override void destroy() {
         foreach (Gst.Element output in outputs) {
             output_tee.unlink(output);
         }
         base.destroy();
+        rotate.set_locked_state(true);
+        rotate.set_state(Gst.State.NULL);
+        rotate.unlink(output_tee);
+        pipe.remove(rotate);
+        rotate = null;
         output_tee.set_locked_state(true);
         output_tee.set_state(Gst.State.NULL);
         pipe.remove(output_tee);
         output_tee = null;
+        disconnect(video_orientation_changed_handler);
     }
 
     public override void add_output(Gst.Element element) {
-        if (element == output_tee) {
+        if (element == output_tee || element == rotate) {
             base.add_output(element);
             return;
         }
@@ -491,7 +669,7 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
     }
 
     public override void remove_output(Gst.Element element) {
-        if (element == output_tee) {
+        if (element == output_tee || element == rotate) {
             base.remove_output(element);
             return;
         }
diff --git a/plugins/rtp/src/video_widget.vala b/plugins/rtp/src/video_widget.vala
index fa5ba138..351069a7 100644
--- a/plugins/rtp/src/video_widget.vala
+++ b/plugins/rtp/src/video_widget.vala
@@ -19,7 +19,7 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         this.plugin = plugin;
 
         id = last_id++;
-        element = Gst.ElementFactory.make("gtksink", @"video-widget-$id");
+        element = Gst.ElementFactory.make("gtksink", @"video_widget_$id");
         if (element != null) {
             Gtk.Widget widget;
             element.@get("widget", out widget);
@@ -51,8 +51,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         if (connected_stream == null) return;
         plugin.pause();
         pipe.add(element);
-        convert = Gst.parse_bin_from_description(@"videoconvert name=video-widget-$id-convert", true);
-        convert.name = @"video-widget-$id-prepare";
+        convert = Gst.parse_bin_from_description(@"videoconvert name=video_widget_$(id)_convert", true);
+        convert.name = @"video_widget_$(id)_prepare";
         pipe.add(convert);
         convert.link(element);
         connected_stream.add_output(convert);
@@ -68,8 +68,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         if (connected_device == null) return;
         plugin.pause();
         pipe.add(element);
-        convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video-widget-$id-flip ! videoconvert name=video-widget-$id-convert", true);
-        convert.name = @"video-widget-$id-prepare";
+        convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video_widget_$(id)_flip ! videoconvert name=video_widget_$(id)_convert", true);
+        convert.name = @"video_widget_$(id)_prepare";
         pipe.add(convert);
         convert.link(element);
         connected_device.link_source().link(convert);
diff --git a/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
new file mode 100644
index 00000000..30490896
--- /dev/null
+++ b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
@@ -0,0 +1,625 @@
+// Fixme: This is fetched from development code of Vala upstream which fixed a few bugs.
+/* gstreamer-rtp-1.0.vapi generated by vapigen, do not modify. */
+
+[CCode (cprefix = "Gst", gir_namespace = "GstRtp", gir_version = "1.0", lower_case_cprefix = "gst_")]
+namespace Gst {
+	namespace RTCP {
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTCPBuffer")]
+		public struct Buffer {
+			public weak Gst.Buffer buffer;
+			public bool add_packet (Gst.RTCP.Type type, Gst.RTCP.Packet packet);
+			public bool get_first_packet (Gst.RTCP.Packet packet);
+			public uint get_packet_count ();
+			public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTCP.Buffer rtcp);
+			public static Gst.Buffer @new (uint mtu);
+			public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] owned uint8[] data);
+			public bool unmap ();
+			public static bool validate (Gst.Buffer buffer);
+			public static bool validate_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			[Version (since = "1.6")]
+			public static bool validate_data_reduced ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			[Version (since = "1.6")]
+			public static bool validate_reduced (Gst.Buffer buffer);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTCPPacket")]
+		public struct Packet {
+			public weak Gst.RTCP.Buffer? rtcp;
+			public uint offset;
+			[Version (since = "1.10")]
+			public bool add_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			public bool add_rb (uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+			[Version (since = "1.10")]
+			public uint8 app_get_data ();
+			[Version (since = "1.10")]
+			public uint16 app_get_data_length ();
+			[Version (since = "1.10")]
+			public unowned string app_get_name ();
+			[Version (since = "1.10")]
+			public uint32 app_get_ssrc ();
+			[Version (since = "1.10")]
+			public uint8 app_get_subtype ();
+			[Version (since = "1.10")]
+			public bool app_set_data_length (uint16 wordlen);
+			[Version (since = "1.10")]
+			public void app_set_name (string name);
+			[Version (since = "1.10")]
+			public void app_set_ssrc (uint32 ssrc);
+			[Version (since = "1.10")]
+			public void app_set_subtype (uint8 subtype);
+			public bool bye_add_ssrc (uint32 ssrc);
+			public bool bye_add_ssrcs ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint32[] ssrc);
+			public uint32 bye_get_nth_ssrc (uint nth);
+			public string bye_get_reason ();
+			public uint8 bye_get_reason_len ();
+			public uint bye_get_ssrc_count ();
+			public bool bye_set_reason (string reason);
+			[Version (since = "1.10")]
+			public bool copy_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out uint8[] data);
+			public uint8 fb_get_fci ();
+			public uint16 fb_get_fci_length ();
+			public uint32 fb_get_media_ssrc ();
+			public uint32 fb_get_sender_ssrc ();
+			public Gst.RTCP.FBType fb_get_type ();
+			public bool fb_set_fci_length (uint16 wordlen);
+			public void fb_set_media_ssrc (uint32 ssrc);
+			public void fb_set_sender_ssrc (uint32 ssrc);
+			public void fb_set_type (Gst.RTCP.FBType type);
+			public uint8 get_count ();
+			public uint16 get_length ();
+			public bool get_padding ();
+			[Version (since = "1.10")]
+			public bool get_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out unowned uint8[] data);
+			[Version (since = "1.10")]
+			public uint16 get_profile_specific_ext_length ();
+			public void get_rb (uint nth, out uint32 ssrc, out uint8 fractionlost, out int32 packetslost, out uint32 exthighestseq, out uint32 jitter, out uint32 lsr, out uint32 dlsr);
+			public uint get_rb_count ();
+			public Gst.RTCP.Type get_type ();
+			public bool move_to_next ();
+			public bool remove ();
+			public uint32 rr_get_ssrc ();
+			public void rr_set_ssrc (uint32 ssrc);
+			public bool sdes_add_entry (Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] uint8[] data);
+			public bool sdes_add_item (uint32 ssrc);
+			public bool sdes_copy_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out uint8[] data);
+			public bool sdes_first_entry ();
+			public bool sdes_first_item ();
+			public bool sdes_get_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out unowned uint8[] data);
+			public uint sdes_get_item_count ();
+			public uint32 sdes_get_ssrc ();
+			public bool sdes_next_entry ();
+			public bool sdes_next_item ();
+			public void set_rb (uint nth, uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+			public void sr_get_sender_info (out uint32 ssrc, out uint64 ntptime, out uint32 rtptime, out uint32 packet_count, out uint32 octet_count);
+			public void sr_set_sender_info (uint32 ssrc, uint64 ntptime, uint32 rtptime, uint32 packet_count, uint32 octet_count);
+			[Version (since = "1.16")]
+			public bool xr_first_rb ();
+			[Version (since = "1.16")]
+			public uint16 xr_get_block_length ();
+			[Version (since = "1.16")]
+			public Gst.RTCP.XRType xr_get_block_type ();
+			[Version (since = "1.16")]
+			public bool xr_get_dlrr_block (uint nth, out uint32 ssrc, out uint32 last_rr, out uint32 delay);
+			[Version (since = "1.16")]
+			public bool xr_get_prt_by_seq (uint16 seq, out uint32 receipt_time);
+			[Version (since = "1.16")]
+			public bool xr_get_prt_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq);
+			[Version (since = "1.16")]
+			public bool xr_get_rle_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq, out uint32 chunk_count);
+			[Version (since = "1.16")]
+			public bool xr_get_rle_nth_chunk (uint nth, out uint16 chunk);
+			[Version (since = "1.16")]
+			public bool xr_get_rrt (out uint64 timestamp);
+			[Version (since = "1.16")]
+			public uint32 xr_get_ssrc ();
+			[Version (since = "1.16")]
+			public bool xr_get_summary_info (out uint32 ssrc, out uint16 begin_seq, out uint16 end_seq);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_jitter (out uint32 min_jitter, out uint32 max_jitter, out uint32 mean_jitter, out uint32 dev_jitter);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_pkt (out uint32 lost_packets, out uint32 dup_packets);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_ttl (out bool is_ipv4, out uint8 min_ttl, out uint8 max_ttl, out uint8 mean_ttl, out uint8 dev_ttl);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_burst_metrics (out uint8 burst_density, out uint8 gap_density, out uint16 burst_duration, out uint16 gap_duration);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_configuration_params (out uint8 gmin, out uint8 rx_config);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_delay_metrics (out uint16 roundtrip_delay, out uint16 end_system_delay);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_jitter_buffer_params (out uint16 jb_nominal, out uint16 jb_maximum, out uint16 jb_abs_max);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_metrics_ssrc (out uint32 ssrc);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_packet_metrics (out uint8 loss_rate, out uint8 discard_rate);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_quality_metrics (out uint8 r_factor, out uint8 ext_r_factor, out uint8 mos_lq, out uint8 mos_cq);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_signal_metrics (out uint8 signal_level, out uint8 noise_level, out uint8 rerl, out uint8 gmin);
+			[Version (since = "1.16")]
+			public bool xr_next_rb ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_", type_id = "gst_rtcpfb_type_get_type ()")]
+		[GIR (name = "RTCPFBType")]
+		public enum FBType {
+			FB_TYPE_INVALID,
+			RTPFB_TYPE_NACK,
+			RTPFB_TYPE_TMMBR,
+			RTPFB_TYPE_TMMBN,
+			RTPFB_TYPE_RTCP_SR_REQ,
+			RTPFB_TYPE_TWCC,
+			PSFB_TYPE_PLI,
+			PSFB_TYPE_SLI,
+			PSFB_TYPE_RPSI,
+			PSFB_TYPE_AFB,
+			PSFB_TYPE_FIR,
+			PSFB_TYPE_TSTR,
+			PSFB_TYPE_TSTN,
+			PSFB_TYPE_VBCN
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_SDES_", type_id = "gst_rtcpsdes_type_get_type ()")]
+		[GIR (name = "RTCPSDESType")]
+		public enum SDESType {
+			INVALID,
+			END,
+			CNAME,
+			NAME,
+			EMAIL,
+			PHONE,
+			LOC,
+			TOOL,
+			NOTE,
+			PRIV;
+			[CCode (cname = "gst_rtcp_sdes_name_to_type")]
+			public static Gst.RTCP.SDESType from_string (string name);
+			[CCode (cname = "gst_rtcp_sdes_type_to_name")]
+			public unowned string to_string ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_TYPE_", type_id = "gst_rtcp_type_get_type ()")]
+		[GIR (name = "RTCPType")]
+		public enum Type {
+			INVALID,
+			SR,
+			RR,
+			SDES,
+			BYE,
+			APP,
+			RTPFB,
+			PSFB,
+			XR
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_XR_TYPE_", type_id = "gst_rtcpxr_type_get_type ()")]
+		[GIR (name = "RTCPXRType")]
+		[Version (since = "1.16")]
+		public enum XRType {
+			INVALID,
+			LRLE,
+			DRLE,
+			PRT,
+			RRT,
+			DLRR,
+			SSUMM,
+			VOIP_METRICS
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_BYE_SSRC_COUNT")]
+		public const int MAX_BYE_SSRC_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_RB_COUNT")]
+		public const int MAX_RB_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES")]
+		public const int MAX_SDES;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES_ITEM_COUNT")]
+		public const int MAX_SDES_ITEM_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_REDUCED_SIZE_VALID_MASK")]
+		public const int REDUCED_SIZE_VALID_MASK;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_MASK")]
+		public const int VALID_MASK;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_VALUE")]
+		public const int VALID_VALUE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VERSION")]
+		public const int VERSION;
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static uint64 ntp_to_unix (uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static uint64 unix_to_ntp (uint64 unixtime);
+	}
+	namespace RTP {
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_audio_payload_get_type ()")]
+		[GIR (name = "RTPBaseAudioPayload")]
+		public class BaseAudioPayload : Gst.RTP.BasePayload {
+			public Gst.ClockTime base_ts;
+			public int frame_duration;
+			public int frame_size;
+			public int sample_size;
+			[CCode (has_construct_function = false)]
+			protected BaseAudioPayload ();
+			public Gst.FlowReturn flush (uint payload_len, Gst.ClockTime timestamp);
+			public Gst.Base.Adapter get_adapter ();
+			public Gst.FlowReturn push ([CCode (array_length_cname = "payload_len", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, Gst.ClockTime timestamp);
+			public void set_frame_based ();
+			public void set_frame_options (int frame_duration, int frame_size);
+			public void set_sample_based ();
+			public void set_sample_options (int sample_size);
+			public void set_samplebits_options (int sample_size);
+			[NoAccessorMethod]
+			public bool buffer_list { get; set; }
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_depayload_get_type ()")]
+		[GIR (name = "RTPBaseDepayload")]
+		public abstract class BaseDepayload : Gst.Element {
+			public uint clock_rate;
+			public bool need_newsegment;
+			public weak Gst.Segment segment;
+			public weak Gst.Pad sinkpad;
+			public weak Gst.Pad srcpad;
+			[CCode (has_construct_function = false)]
+			protected BaseDepayload ();
+			[NoWrapper]
+			public virtual bool handle_event (Gst.Event event);
+			[Version (since = "1.16")]
+			public bool is_source_info_enabled ();
+			[NoWrapper]
+			public virtual bool packet_lost (Gst.Event event);
+			[NoWrapper]
+			public virtual Gst.Buffer process (Gst.Buffer @in);
+			[NoWrapper]
+			public virtual Gst.Buffer process_rtp_packet (Gst.RTP.Buffer rtp_buffer);
+			public Gst.FlowReturn push (Gst.Buffer out_buf);
+			public Gst.FlowReturn push_list (Gst.BufferList out_list);
+			[NoWrapper]
+			public virtual bool set_caps (Gst.Caps caps);
+			[Version (since = "1.16")]
+			public void set_source_info_enabled (bool enable);
+			[NoAccessorMethod]
+			[Version (since = "1.20")]
+			public bool auto_header_extension { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.18")]
+			public int max_reorder { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool source_info { get; set; }
+			[NoAccessorMethod]
+			public Gst.Structure stats { owned get; }
+			[Version (since = "1.20")]
+			public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+			[Version (since = "1.20")]
+			public signal void clear_extensions ();
+			[Version (since = "1.20")]
+			public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string? ext_uri);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_payload_get_type ()")]
+		[GIR (name = "RTPBasePayload")]
+		public abstract class BasePayload : Gst.Element {
+			[CCode (has_construct_function = false)]
+			protected BasePayload ();
+			[Version (since = "1.16")]
+			public Gst.Buffer allocate_output_buffer (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			[NoWrapper]
+			public virtual Gst.Caps get_caps (Gst.Pad pad, Gst.Caps filter);
+			[Version (since = "1.16")]
+			public uint get_source_count (Gst.Buffer buffer);
+			[NoWrapper]
+			public virtual Gst.FlowReturn handle_buffer (Gst.Buffer buffer);
+			public bool is_filled (uint size, Gst.ClockTime duration);
+			[Version (since = "1.16")]
+			public bool is_source_info_enabled ();
+			public Gst.FlowReturn push (Gst.Buffer buffer);
+			public Gst.FlowReturn push_list (Gst.BufferList list);
+			[NoWrapper]
+			public virtual bool query (Gst.Pad pad, Gst.Query query);
+			[NoWrapper]
+			public virtual bool set_caps (Gst.Caps caps);
+			public void set_options (string media, bool @dynamic, string encoding_name, uint32 clock_rate);
+			[Version (since = "1.20")]
+			public bool set_outcaps_structure (Gst.Structure? s);
+			[Version (since = "1.16")]
+			public void set_source_info_enabled (bool enable);
+			[NoWrapper]
+			public virtual bool sink_event (Gst.Event event);
+			[NoWrapper]
+			public virtual bool src_event (Gst.Event event);
+			[NoAccessorMethod]
+			[Version (since = "1.20")]
+			public bool auto_header_extension { get; set; }
+			[NoAccessorMethod]
+			public int64 max_ptime { get; set; }
+			[NoAccessorMethod]
+			public int64 min_ptime { get; set; }
+			[NoAccessorMethod]
+			public uint mtu { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool onvif_no_rate_control { get; set; }
+			[NoAccessorMethod]
+			public bool perfect_rtptime { get; set; }
+			[NoAccessorMethod]
+			public uint pt { get; set; }
+			[NoAccessorMethod]
+			public int64 ptime_multiple { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.18")]
+			public bool scale_rtptime { get; set; }
+			[NoAccessorMethod]
+			public uint seqnum { get; }
+			[NoAccessorMethod]
+			public int seqnum_offset { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool source_info { get; set; }
+			[NoAccessorMethod]
+			public uint ssrc { get; set; }
+			[NoAccessorMethod]
+			public Gst.Structure stats { owned get; }
+			[NoAccessorMethod]
+			public uint timestamp { get; }
+			[NoAccessorMethod]
+			public uint timestamp_offset { get; set; }
+			[Version (since = "1.20")]
+			public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+			[Version (since = "1.20")]
+			public signal void clear_extensions ();
+			[Version (since = "1.20")]
+			public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string ext_uri);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_header_extension_get_type ()")]
+		[GIR (name = "RTPHeaderExtension")]
+		[Version (since = "1.20")]
+		public abstract class HeaderExtension : Gst.Element {
+			public uint ext_id;
+			[CCode (has_construct_function = false)]
+			protected HeaderExtension ();
+			public static Gst.RTP.HeaderExtension? create_from_uri (string uri);
+			public uint get_id ();
+			public virtual size_t get_max_size (Gst.Buffer input_meta);
+			public string get_sdp_caps_field_name ();
+			public virtual Gst.RTP.HeaderExtensionFlags get_supported_flags ();
+			public unowned string get_uri ();
+			public virtual bool read (Gst.RTP.HeaderExtensionFlags read_flags, [CCode (array_length_cname = "size", array_length_pos = 2.5, array_length_type = "gsize", type = "const guint8*")] uint8[] data, Gst.Buffer buffer);
+			public virtual bool set_attributes_from_caps (Gst.Caps caps);
+			public bool set_attributes_from_caps_simple_sdp (Gst.Caps caps);
+			public virtual bool set_caps_from_attributes (Gst.Caps caps);
+			public bool set_caps_from_attributes_simple_sdp (Gst.Caps caps);
+			public void set_id (uint ext_id);
+			public virtual bool set_non_rtp_sink_caps (Gst.Caps caps);
+			[CCode (cname = "gst_rtp_header_extension_class_set_uri")]
+			public class void set_uri (string uri);
+			public void set_wants_update_non_rtp_src_caps (bool state);
+			public virtual bool update_non_rtp_src_caps (Gst.Caps caps);
+			public virtual size_t write (Gst.Buffer input_meta, Gst.RTP.HeaderExtensionFlags write_flags, Gst.Buffer output, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "gsize", type = "guint8*")] uint8[] data);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPBuffer")]
+		public struct Buffer {
+			public weak Gst.Buffer buffer;
+			public uint state;
+			[CCode (array_length = false)]
+			public weak void* data[4];
+			[CCode (array_length = false)]
+			public weak size_t size[4];
+			public bool add_extension_onebyte_header (uint8 id, [CCode (array_length_cname = "size", array_length_pos = 2.1, array_length_type = "guint")] uint8[] data);
+			public bool add_extension_twobytes_header (uint8 appbits, uint8 id, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] uint8[] data);
+			[CCode (cname = "gst_buffer_add_rtp_source_meta")]
+			[Version (since = "1.16")]
+			public static unowned Gst.RTP.SourceMeta? add_rtp_source_meta (Gst.Buffer buffer, uint32? ssrc, uint32? csrc, uint csrc_count);
+			public static void allocate_data (Gst.Buffer buffer, uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static uint calc_header_len (uint8 csrc_count);
+			public static uint calc_packet_len (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static uint calc_payload_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+			public static int compare_seqnum (uint16 seqnum1, uint16 seqnum2);
+			public static uint32 default_clock_rate (uint8 payload_type);
+			public static uint64 ext_timestamp (ref uint64 exttimestamp, uint32 timestamp);
+			public uint32 get_csrc (uint8 idx);
+			public uint8 get_csrc_count ();
+			public bool get_extension ();
+			[Version (since = "1.2")]
+			public GLib.Bytes get_extension_bytes (out uint16 bits);
+			public bool get_extension_data (out uint16 bits, [CCode (array_length = false)] out unowned uint8[] data, out uint wordlen);
+			public bool get_extension_onebyte_header (uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] out unowned uint8[] data);
+			[Version (since = "1.18")]
+			public static bool get_extension_onebyte_header_from_bytes (GLib.Bytes bytes, uint16 bit_pattern, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 5.1, array_length_type = "guint")] out unowned uint8[] data);
+			public bool get_extension_twobytes_header (out uint8 appbits, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "guint")] out unowned uint8[] data);
+			public uint get_header_len ();
+			public bool get_marker ();
+			public uint get_packet_len ();
+			public bool get_padding ();
+			[CCode (array_length = false)]
+			public unowned uint8[] get_payload ();
+			public Gst.Buffer get_payload_buffer ();
+			[Version (since = "1.2")]
+			public GLib.Bytes get_payload_bytes ();
+			public uint get_payload_len ();
+			public Gst.Buffer get_payload_subbuffer (uint offset, uint len);
+			public uint8 get_payload_type ();
+			[CCode (cname = "gst_buffer_get_rtp_source_meta")]
+			[Version (since = "1.16")]
+			public static unowned Gst.RTP.SourceMeta? get_rtp_source_meta (Gst.Buffer buffer);
+			public uint16 get_seq ();
+			public uint32 get_ssrc ();
+			public uint32 get_timestamp ();
+			public uint8 get_version ();
+			public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTP.Buffer rtp);
+			public static Gst.Buffer new_allocate (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static Gst.Buffer new_allocate_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+			public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] uint8[] data);
+			public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] owned uint8[] data);
+			public void pad_to (uint len);
+			public void set_csrc (uint8 idx, uint32 csrc);
+			public void set_extension (bool extension);
+			public bool set_extension_data (uint16 bits, uint16 length);
+			public void set_marker (bool marker);
+			public void set_packet_len (uint len);
+			public void set_padding (bool padding);
+			public void set_payload_type (uint8 payload_type);
+			public void set_seq (uint16 seq);
+			public void set_ssrc (uint32 ssrc);
+			public void set_timestamp (uint32 timestamp);
+			public void set_version (uint8 version);
+			public void unmap ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPPayloadInfo")]
+		public struct PayloadInfo {
+			public uint8 payload_type;
+			public weak string media;
+			public weak string encoding_name;
+			public uint clock_rate;
+			public weak string encoding_parameters;
+			public uint bitrate;
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPSourceMeta")]
+		[Version (since = "1.16")]
+		public struct SourceMeta {
+			public Gst.Meta meta;
+			public uint32 ssrc;
+			public bool ssrc_valid;
+			[CCode (array_length = false)]
+			public weak uint32 csrc[15];
+			public uint csrc_count;
+			public bool append_csrc ([CCode (array_length_cname = "csrc_count", array_length_pos = 1.1, array_length_type = "guint", type = "const guint32*")] uint32[] csrc);
+			public uint get_source_count ();
+			public bool set_ssrc (uint32? ssrc);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_FLAG_", type_id = "gst_rtp_buffer_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPBufferFlags")]
+		[Version (since = "1.10")]
+		public enum BufferFlags {
+			RETRANSMISSION,
+			REDUNDANT,
+			LAST
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_MAP_FLAG_", type_id = "gst_rtp_buffer_map_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPBufferMapFlags")]
+		[Version (since = "1.6.1")]
+		public enum BufferMapFlags {
+			SKIP_PADDING,
+			LAST
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_HEADER_EXTENSION_", type_id = "gst_rtp_header_extension_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPHeaderExtensionFlags")]
+		[Version (since = "1.20")]
+		public enum HeaderExtensionFlags {
+			ONE_BYTE,
+			TWO_BYTE
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PAYLOAD_", type_id = "gst_rtp_payload_get_type ()")]
+		[GIR (name = "RTPPayload")]
+		public enum Payload {
+			PCMU,
+			@1016,
+			G721,
+			GSM,
+			G723,
+			DVI4_8000,
+			DVI4_16000,
+			LPC,
+			PCMA,
+			G722,
+			L16_STEREO,
+			L16_MONO,
+			QCELP,
+			CN,
+			MPA,
+			G728,
+			DVI4_11025,
+			DVI4_22050,
+			G729,
+			CELLB,
+			JPEG,
+			NV,
+			H261,
+			MPV,
+			MP2T,
+			H263;
+			public const string @1016_STRING;
+			public const string CELLB_STRING;
+			public const string CN_STRING;
+			public const string DVI4_11025_STRING;
+			public const string DVI4_16000_STRING;
+			public const string DVI4_22050_STRING;
+			public const string DVI4_8000_STRING;
+			public const string DYNAMIC_STRING;
+			public const string G721_STRING;
+			public const string G722_STRING;
+			public const int G723_53;
+			public const string G723_53_STRING;
+			public const int G723_63;
+			public const string G723_63_STRING;
+			public const string G723_STRING;
+			public const string G728_STRING;
+			public const string G729_STRING;
+			public const string GSM_STRING;
+			public const string H261_STRING;
+			public const string H263_STRING;
+			public const string JPEG_STRING;
+			public const string L16_MONO_STRING;
+			public const string L16_STEREO_STRING;
+			public const string LPC_STRING;
+			public const string MP2T_STRING;
+			public const string MPA_STRING;
+			public const string MPV_STRING;
+			public const string NV_STRING;
+			public const string PCMA_STRING;
+			public const string PCMU_STRING;
+			public const string QCELP_STRING;
+			public const int TS41;
+			public const string TS41_STRING;
+			public const int TS48;
+			public const string TS48_STRING;
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PROFILE_", type_id = "gst_rtp_profile_get_type ()")]
+		[GIR (name = "RTPProfile")]
+		[Version (since = "1.6")]
+		public enum Profile {
+			UNKNOWN,
+			AVP,
+			SAVP,
+			AVPF,
+			SAVPF
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_BASE")]
+		public const string HDREXT_BASE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_ELEMENT_CLASS")]
+		[Version (since = "1.20")]
+		public const string HDREXT_ELEMENT_CLASS;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56")]
+		public const string HDREXT_NTP_56;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56_SIZE")]
+		public const int HDREXT_NTP_56_SIZE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64")]
+		public const string HDREXT_NTP_64;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64_SIZE")]
+		public const int HDREXT_NTP_64_SIZE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HEADER_EXTENSION_URI_METADATA_KEY")]
+		[Version (since = "1.20")]
+		public const string HEADER_EXTENSION_URI_METADATA_KEY;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_SOURCE_META_MAX_CSRC_COUNT")]
+		public const int SOURCE_META_MAX_CSRC_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_VERSION")]
+		public const int VERSION;
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		[Version (since = "1.20")]
+		public static GLib.List<Gst.RTP.HeaderExtension> get_header_extension_list ();
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_get_ntp_56 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_get_ntp_64 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_set_ntp_56 (void* data, uint size, uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_set_ntp_64 (void* data, uint size, uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.RTP.PayloadInfo? payload_info_for_name (string media, string encoding_name);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.RTP.PayloadInfo? payload_info_for_pt (uint8 payload_type);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static GLib.Type source_meta_api_get_type ();
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.MetaInfo? source_meta_get_info ();
+	}
+}
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
index d6f1acd2..d4440169 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/content_parameters.vala
@@ -17,6 +17,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
     public bool encryption_required { get; private set; default = false; }
     public PayloadType? agreed_payload_type { get; private set; }
     public Gee.List<PayloadType> payload_types = new ArrayList<PayloadType>(PayloadType.equals_func);
+    public Gee.List<HeaderExtension> header_extensions = new ArrayList<HeaderExtension>();
     public Gee.List<Crypto> remote_cryptos = new ArrayList<Crypto>();
     public Crypto? local_crypto = null;
     public Crypto? remote_crypto = null;
@@ -54,9 +55,12 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
                 this.remote_cryptos.add(Crypto.parse(crypto));
             }
         }
-        foreach (StanzaNode payloadType in node.get_subnodes("payload-type")) {
+        foreach (StanzaNode payloadType in node.get_subnodes(PayloadType.NAME)) {
             this.payload_types.add(PayloadType.parse(payloadType));
         }
+        foreach (StanzaNode subnode in node.get_subnodes(HeaderExtension.NAME, HeaderExtension.NS_URI)) {
+            this.header_extensions.add(HeaderExtension.parse(subnode));
+        }
     }
 
     public async void handle_proposed_content(XmppStream stream, Jingle.Session session, Jingle.Content content) {
@@ -66,6 +70,11 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
             content.reject();
             return;
         }
+        // Drop unsupported header extensions
+        var iter = header_extensions.iterator();
+        while(iter.next()) {
+            if (!parent.is_header_extension_supported(media, iter.@get())) iter.remove();
+        }
         remote_crypto = parent.pick_remote_crypto(remote_cryptos);
         if (local_crypto == null && remote_crypto != null) {
             local_crypto = parent.pick_local_crypto(remote_crypto);
@@ -151,7 +160,7 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
 
         Gee.List<StanzaNode> crypto_nodes = description_node.get_deep_subnodes("encryption", "crypto");
         if (crypto_nodes.size == 0) {
-            warning("Counterpart didn't include any cryptos");
+            debug("Counterpart didn't include any cryptos");
             if (encryption_required) {
                 return;
             }
@@ -182,6 +191,9 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
                 ret.put_node(payload_type.to_xml());
             }
         }
+        foreach (HeaderExtension ext in header_extensions) {
+            ret.put_node(ext.to_xml());
+        }
         if (local_crypto != null) {
             ret.put_node(new StanzaNode.build("encryption", NS_URI)
                 .put_node(local_crypto.to_xml()));
@@ -191,4 +203,28 @@ public class Xmpp.Xep.JingleRtp.Parameters : Jingle.ContentParameters, Object {
         }
         return ret;
     }
+}
+
+public class Xmpp.Xep.JingleRtp.HeaderExtension {
+    public const string NS_URI = "urn:xmpp:jingle:apps:rtp:rtp-hdrext:0";
+    public const string NAME = "rtp-hdrext";
+
+    public uint8 id { get; private set; }
+    public string uri { get; private set; }
+
+    public HeaderExtension(uint8 id, string uri) {
+        this.id = id;
+        this.uri = uri;
+    }
+
+    public static HeaderExtension parse(StanzaNode node) {
+        return new HeaderExtension((uint8) node.get_attribute_int("id"), node.get_attribute("uri"));
+    }
+
+    public StanzaNode to_xml() {
+        return new StanzaNode.build(NAME, NS_URI)
+                .add_self_xmlns()
+                .put_attribute("id", id.to_string())
+                .put_attribute("uri", uri);
+    }
 }
\ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
index 3adad114..6eb6289b 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/jingle_rtp_module.vala
@@ -24,6 +24,8 @@ public abstract class Module : XmppStreamModule {
     public abstract Crypto? pick_remote_crypto(Gee.List<Crypto> cryptos);
     public abstract Crypto? pick_local_crypto(Crypto? remote);
     public abstract Stream create_stream(Jingle.Content content);
+    public abstract bool is_header_extension_supported(string media, HeaderExtension ext);
+    public abstract Gee.List<HeaderExtension> get_suggested_header_extensions(string media);
     public abstract void close_stream(Stream stream);
 
     public async Jingle.Session start_call(XmppStream stream, Jid receiver_full_jid, bool video, string? sid = null) throws Jingle.Error {
@@ -40,6 +42,7 @@ public abstract class Module : XmppStreamModule {
         // Create audio content
         Parameters audio_content_parameters = new Parameters(this, "audio", yield get_supported_payloads("audio"));
         audio_content_parameters.local_crypto = generate_local_crypto();
+        audio_content_parameters.header_extensions.add_all(get_suggested_header_extensions("audio"));
         Jingle.Transport? audio_transport = yield jingle_module.select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
         if (audio_transport == null) {
             throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable audio transports");
@@ -57,6 +60,7 @@ public abstract class Module : XmppStreamModule {
             // Create video content
             Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
             video_content_parameters.local_crypto = generate_local_crypto();
+            video_content_parameters.header_extensions.add_all(get_suggested_header_extensions("video"));
             Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
             if (video_transport == null) {
                 throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
@@ -98,6 +102,7 @@ public abstract class Module : XmppStreamModule {
             // Content for video does not yet exist -> create it
             Parameters video_content_parameters = new Parameters(this, "video", yield get_supported_payloads("video"));
             video_content_parameters.local_crypto = generate_local_crypto();
+            video_content_parameters.header_extensions.add_all(get_suggested_header_extensions("video"));
             Jingle.Transport? video_transport = yield stream.get_module(Jingle.Module.IDENTITY).select_transport(stream, content_type.required_transport_type, content_type.required_components, receiver_full_jid, Set.empty());
             if (video_transport == null) {
                 throw new Jingle.Error.NO_SHARED_PROTOCOLS("No suitable video transports");
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
index 452f1d65..faba38c9 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/payload_type.vala
@@ -3,6 +3,8 @@ using Xmpp;
 using Xmpp.Xep;
 
 public class Xmpp.Xep.JingleRtp.PayloadType {
+    public const string NAME = "payload-type";
+
     public uint8 id { get; set; }
     public string? name { get; set; }
     public uint8 channels { get; set; default = 1; }
@@ -10,6 +12,7 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
     public uint32 maxptime { get; set; }
     public uint32 ptime { get; set; }
     public Map<string, string> parameters = new HashMap<string, string>();
+    public Gee.List<RtcpFeedback> rtcp_fbs = new ArrayList<RtcpFeedback>();
 
     public static PayloadType parse(StanzaNode node) {
         PayloadType payloadType = new PayloadType();
@@ -22,11 +25,14 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
         foreach (StanzaNode parameter in node.get_subnodes("parameter")) {
             payloadType.parameters[parameter.get_attribute("name")] = parameter.get_attribute("value");
         }
+        foreach (StanzaNode subnode in node.get_subnodes(RtcpFeedback.NAME, RtcpFeedback.NS_URI)) {
+            payloadType.rtcp_fbs.add(RtcpFeedback.parse(subnode));
+        }
         return payloadType;
     }
 
     public StanzaNode to_xml() {
-        StanzaNode node = new StanzaNode.build("payload-type", NS_URI)
+        StanzaNode node = new StanzaNode.build(NAME, NS_URI)
                 .put_attribute("id", id.to_string());
         if (channels != 1) node.put_attribute("channels", channels.to_string());
         if (clockrate != 0) node.put_attribute("clockrate", clockrate.to_string());
@@ -38,9 +44,25 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
                     .put_attribute("name", parameter)
                     .put_attribute("value", parameters[parameter]));
         }
+        foreach (RtcpFeedback rtcp_fb in rtcp_fbs) {
+            node.put_node(rtcp_fb.to_xml());
+        }
         return node;
     }
 
+    public PayloadType clone() {
+        PayloadType clone = new PayloadType();
+        clone.id = id;
+        clone.name = name;
+        clone.channels = channels;
+        clone.clockrate = clockrate;
+        clone.maxptime = maxptime;
+        clone.ptime = ptime;
+        clone.parameters.set_all(parameters);
+        clone.rtcp_fbs.add_all(rtcp_fbs);
+        return clone;
+    }
+
     public static bool equals_func(PayloadType a, PayloadType b) {
         return a.id == b.id &&
                 a.name == b.name &&
@@ -49,4 +71,29 @@ public class Xmpp.Xep.JingleRtp.PayloadType {
                 a.maxptime == b.maxptime &&
                 a.ptime == b.ptime;
     }
+}
+
+public class Xmpp.Xep.JingleRtp.RtcpFeedback {
+    public const string NS_URI = "urn:xmpp:jingle:apps:rtp:rtcp-fb:0";
+    public const string NAME = "rtcp-fb";
+
+    public string type_ { get; private set; }
+    public string? subtype { get; private set; }
+
+    public RtcpFeedback(string type, string? subtype = null) {
+        this.type_ = type;
+        this.subtype = subtype;
+    }
+
+    public static RtcpFeedback parse(StanzaNode node) {
+        return new RtcpFeedback(node.get_attribute("type"), node.get_attribute("subtype"));
+    }
+
+    public StanzaNode to_xml() {
+        StanzaNode node = new StanzaNode.build(NAME, NS_URI)
+                .add_self_xmlns()
+                .put_attribute("type", type_);
+        if (subtype != null) node.put_attribute("subtype", subtype);
+        return node;
+    }
 }
\ No newline at end of file
diff --git a/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala b/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
index adae11f5..65be8a0a 100644
--- a/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
+++ b/xmpp-vala/src/module/xep/0167_jingle_rtp/stream.vala
@@ -33,6 +33,13 @@ public abstract class Xmpp.Xep.JingleRtp.Stream : Object {
         }
         return null;
     }}
+    public Gee.List<JingleRtp.HeaderExtension>? header_extensions { get {
+        var content_params = content.content_params;
+        if (content_params is Parameters) {
+            return ((Parameters)content_params).header_extensions;
+        }
+        return null;
+    }}
     public bool sending { get {
         return content.session.senders_include_us(content.senders);
     }}
-- 
cgit v1.2.3-70-g09d2


From 23ffd37dded3bf872e42d7a00727ab3c4d105a97 Mon Sep 17 00:00:00 2001
From: Marvin W <git@larma.de>
Date: Sat, 1 May 2021 15:19:05 +0200
Subject: Echo Cancellation

---
 CMakeLists.txt                             |   6 +-
 cmake/FindGstAudio.cmake                   |  14 +++
 cmake/FindWebRTCAudioProcessing.cmake      |  12 ++
 plugins/rtp/CMakeLists.txt                 |  20 +++-
 plugins/rtp/src/device.vala                |  30 +++--
 plugins/rtp/src/plugin.vala                |   5 +-
 plugins/rtp/src/voice_processor.vala       | 176 +++++++++++++++++++++++++++++
 plugins/rtp/src/voice_processor_native.cpp | 141 +++++++++++++++++++++++
 8 files changed, 385 insertions(+), 19 deletions(-)
 create mode 100644 cmake/FindGstAudio.cmake
 create mode 100644 cmake/FindWebRTCAudioProcessing.cmake
 create mode 100644 plugins/rtp/src/voice_processor.vala
 create mode 100644 plugins/rtp/src/voice_processor_native.cpp

(limited to 'cmake')

diff --git a/CMakeLists.txt b/CMakeLists.txt
index f480b0b2..b3bd35cc 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -2,11 +2,11 @@ cmake_minimum_required(VERSION 3.3)
 list(APPEND CMAKE_MODULE_PATH ${CMAKE_SOURCE_DIR}/cmake)
 include(ComputeVersion)
 if (NOT VERSION_FOUND)
-    project(Dino LANGUAGES C)
+    project(Dino LANGUAGES C CXX)
 elseif (VERSION_IS_RELEASE)
-    project(Dino VERSION ${VERSION_FULL} LANGUAGES C)
+    project(Dino VERSION ${VERSION_FULL} LANGUAGES C CXX)
 else ()
-    project(Dino LANGUAGES C)
+    project(Dino LANGUAGES C CXX)
     set(PROJECT_VERSION ${VERSION_FULL})
 endif ()
 
diff --git a/cmake/FindGstAudio.cmake b/cmake/FindGstAudio.cmake
new file mode 100644
index 00000000..d5fc5dfb
--- /dev/null
+++ b/cmake/FindGstAudio.cmake
@@ -0,0 +1,14 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(GstAudio
+    PKG_CONFIG_NAME gstreamer-audio-1.0
+    LIB_NAMES gstaudio
+    LIB_DIR_HINTS gstreamer-1.0
+    INCLUDE_NAMES gst/audio/audio.h
+    INCLUDE_DIR_SUFFIXES gstreamer-1.0 gstreamer-1.0/include gstreamer-audio-1.0 gstreamer-audio-1.0/include
+    DEPENDS Gst
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstAudio
+    REQUIRED_VARS GstAudio_LIBRARY
+    VERSION_VAR GstAudio_VERSION)
diff --git a/cmake/FindWebRTCAudioProcessing.cmake b/cmake/FindWebRTCAudioProcessing.cmake
new file mode 100644
index 00000000..5f17805d
--- /dev/null
+++ b/cmake/FindWebRTCAudioProcessing.cmake
@@ -0,0 +1,12 @@
+include(PkgConfigWithFallback)
+find_pkg_config_with_fallback(WebRTCAudioProcessing
+    PKG_CONFIG_NAME webrtc-audio-processing
+    LIB_NAMES webrtc_audio_processing
+    INCLUDE_NAMES webrtc/modules/audio_processing/include/audio_processing.h
+    INCLUDE_DIR_SUFFIXES webrtc-audio-processing webrtc_audio_processing
+)
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(WebRTCAudioProcessing
+    REQUIRED_VARS WebRTCAudioProcessing_LIBRARY
+    VERSION_VAR WebRTCAudioProcessing_VERSION)
diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 92ec1b97..b19c8a8f 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -1,4 +1,5 @@
 find_package(GstRtp REQUIRED)
+find_package(WebRTCAudioProcessing 0.2)
 find_packages(RTP_PACKAGES REQUIRED
     Gee
     GLib
@@ -8,12 +9,26 @@ find_packages(RTP_PACKAGES REQUIRED
     GTK3
     Gst
     GstApp
+    GstAudio
 )
 
 if(Gst_VERSION VERSION_GREATER "1.16")
     set(RTP_DEFINITIONS GST_1_16)
 endif()
 
+if(WebRTCAudioProcessing_VERSION GREATER "0.4")
+    message(WARNING "Ignoring WebRTCAudioProcessing, only versions < 0.4 supported so far")
+    unset(WebRTCAudioProcessing_FOUND)
+endif()
+
+if(WebRTCAudioProcessing_FOUND)
+    set(RTP_DEFINITIONS ${RTP_DEFINITIONS} WITH_VOICE_PROCESSOR)
+    set(RTP_VOICE_PROCESSOR_VALA src/voice_processor.vala)
+    set(RTP_VOICE_PROCESSOR_CXX src/voice_processor_native.cpp)
+else()
+    message(WARNING "WebRTCAudioProcessing not found, build without voice pre-processing!")
+endif()
+
 vala_precompile(RTP_VALA_C
 SOURCES
     src/codec_util.vala
@@ -23,6 +38,7 @@ SOURCES
     src/stream.vala
     src/video_widget.vala
     src/register_plugin.vala
+    ${RTP_VOICE_PROCESSOR_VALA}
 CUSTOM_VAPIS
     ${CMAKE_BINARY_DIR}/exports/crypto-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
@@ -36,8 +52,8 @@ DEFINITIONS
 )
 
 add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
-add_library(rtp SHARED ${RTP_VALA_C})
-target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0)
+add_library(rtp SHARED ${RTP_VALA_C} ${RTP_VOICE_PROCESSOR_CXX})
+target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0 webrtc-audio-processing)
 set_target_properties(rtp PROPERTIES PREFIX "")
 set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
 
diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala
index 785f853a..f8894502 100644
--- a/plugins/rtp/src/device.vala
+++ b/plugins/rtp/src/device.vala
@@ -37,6 +37,7 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
     private Gst.Element dsp;
     private Gst.Element mixer;
     private Gst.Element filter;
+    private Gst.Element rate;
     private int links = 0;
 
     public Device(Plugin plugin, Gst.Device device) {
@@ -132,12 +133,10 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             pipe.add(filter);
             element.link(filter);
             if (media == "audio" && plugin.echoprobe != null) {
-                dsp = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
-                if (dsp != null) {
-                    dsp.@set("probe", plugin.echoprobe.name);
-                    pipe.add(dsp);
-                    filter.link(dsp);
-                }
+                dsp = new VoiceProcessor(plugin.echoprobe, element as Gst.Audio.StreamVolume);
+                dsp.name = @"dsp_$id";
+                pipe.add(dsp);
+                filter.link(dsp);
             }
             tee = Gst.ElementFactory.make("tee", @"tee_$id");
             tee.@set("allow-not-linked", true);
@@ -153,7 +152,11 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             if (plugin.echoprobe != null) {
-                filter.link(plugin.echoprobe);
+                rate = Gst.ElementFactory.make("audiorate", @"rate_$id");
+                rate.@set("tolerance", 100000000);
+                pipe.add(rate);
+                filter.link(rate);
+                rate.link(plugin.echoprobe);
                 plugin.echoprobe.link(element);
             } else {
                 filter.link(element);
@@ -184,14 +187,17 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             if (filter != null) {
                 filter.set_locked_state(true);
                 filter.set_state(Gst.State.NULL);
-                if (plugin.echoprobe != null) {
-                    filter.unlink(plugin.echoprobe);
-                } else {
-                    filter.unlink(element);
-                }
+                filter.unlink(rate ?? ((Gst.Element)plugin.echoprobe) ?? element);
                 pipe.remove(filter);
                 filter = null;
             }
+            if (rate != null) {
+                rate.set_locked_state(true);
+                rate.set_state(Gst.State.NULL);
+                rate.unlink(plugin.echoprobe);
+                pipe.remove(rate);
+                rate = null;
+            }
             if (plugin.echoprobe != null) {
                 plugin.echoprobe.unlink(element);
             }
diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala
index d43588b4..e3d5ee41 100644
--- a/plugins/rtp/src/plugin.vala
+++ b/plugins/rtp/src/plugin.vala
@@ -8,7 +8,7 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
     public Gst.DeviceMonitor device_monitor { get; private set; }
     public Gst.Pipeline pipe { get; private set; }
     public Gst.Bin rtpbin { get; private set; }
-    public Gst.Element echoprobe { get; private set; }
+    public EchoProbe echoprobe { get; private set; }
 
     private Gee.List<Stream> streams = new ArrayList<Stream>();
     private Gee.List<Device> devices = new ArrayList<Device>();
@@ -72,7 +72,8 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
         pipe.add(rtpbin);
 
         // Audio echo probe
-        echoprobe = Gst.ElementFactory.make("webrtcechoprobe", "echo-probe");
+//        echoprobe = Gst.ElementFactory.make("webrtcechoprobe", "echo-probe");
+        echoprobe = new EchoProbe();
         if (echoprobe != null) pipe.add(echoprobe);
 
         // Pipeline
diff --git a/plugins/rtp/src/voice_processor.vala b/plugins/rtp/src/voice_processor.vala
new file mode 100644
index 00000000..e6dc7e8f
--- /dev/null
+++ b/plugins/rtp/src/voice_processor.vala
@@ -0,0 +1,176 @@
+using Gst;
+
+namespace Dino.Plugins.Rtp {
+public static extern Buffer adjust_to_running_time(Base.Transform transform, Buffer buf);
+}
+
+public class Dino.Plugins.Rtp.EchoProbe : Audio.Filter {
+    private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
+    private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
+    public Audio.Info audio_info { get; private set; }
+    public signal void on_new_buffer(Buffer buffer);
+    private uint period_samples;
+    private uint period_size;
+    private Base.Adapter adapter = new Base.Adapter();
+
+    static construct {
+        add_static_pad_template(sink_template);
+        add_static_pad_template(src_template);
+        set_static_metadata("Acoustic Echo Canceller probe", "Generic/Audio", "Gathers playback buffers for echo cancellation", "Dino Team <contact@dino.im>");
+    }
+
+    construct {
+        set_passthrough(true);
+    }
+
+    public override bool setup(Audio.Info info) {
+        audio_info = info;
+        period_samples = info.rate / 100; // 10ms buffers
+        period_size = period_samples * info.bpf;
+        return true;
+    }
+
+
+    public override FlowReturn transform_ip(Buffer buf) {
+        lock (adapter) {
+            adapter.push(adjust_to_running_time(this, buf));
+            while (adapter.available() > period_size) {
+                on_new_buffer(adapter.take_buffer(period_size));
+            }
+        }
+        return FlowReturn.OK;
+    }
+
+    public override bool stop() {
+        adapter.clear();
+        return true;
+    }
+}
+
+public class Dino.Plugins.Rtp.VoiceProcessor : Audio.Filter {
+    private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
+    private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
+    public Audio.Info audio_info { get; private set; }
+    private ulong process_outgoing_buffer_handler_id;
+    private uint adjust_delay_timeout_id;
+    private uint period_samples;
+    private uint period_size;
+    private Base.Adapter adapter = new Base.Adapter();
+    private EchoProbe? echo_probe;
+    private Audio.StreamVolume? stream_volume;
+    private ClockTime last_reverse;
+    private void* native;
+
+    static construct {
+        add_static_pad_template(sink_template);
+        add_static_pad_template(src_template);
+        set_static_metadata("Voice Processor (AGC, AEC, filters, etc.)", "Generic/Audio", "Pre-processes voice with WebRTC Audio Processing Library", "Dino Team <contact@dino.im>");
+    }
+
+    construct {
+        set_passthrough(false);
+    }
+
+    public VoiceProcessor(EchoProbe? echo_probe = null, Audio.StreamVolume? stream_volume = null) {
+        this.echo_probe = echo_probe;
+        this.stream_volume = stream_volume;
+    }
+
+    private static extern void* init_native(int stream_delay);
+    private static extern void setup_native(void* native);
+    private static extern void destroy_native(void* native);
+    private static extern void analyze_reverse_stream(void* native, Audio.Info info, Buffer buffer);
+    private static extern void process_stream(void* native, Audio.Info info, Buffer buffer);
+    private static extern void adjust_stream_delay(void* native);
+    private static extern void notify_gain_level(void* native, int gain_level);
+    private static extern int get_suggested_gain_level(void* native);
+    private static extern bool get_stream_has_voice(void* native);
+
+    public override bool setup(Audio.Info info) {
+        debug("VoiceProcessor.setup(%s)", info.to_caps().to_string());
+        audio_info = info;
+        period_samples = info.rate / 100; // 10ms buffers
+        period_size = period_samples * info.bpf;
+        adapter.clear();
+        setup_native(native);
+        return true;
+    }
+
+    public override bool start() {
+        native = init_native(150);
+        if (process_outgoing_buffer_handler_id == 0 && echo_probe != null) {
+            process_outgoing_buffer_handler_id = echo_probe.on_new_buffer.connect(process_outgoing_buffer);
+        }
+        if (stream_volume == null && sinkpad.get_peer() != null && sinkpad.get_peer().get_parent_element() is Audio.StreamVolume) {
+            stream_volume = sinkpad.get_peer().get_parent_element() as Audio.StreamVolume;
+        }
+        return true;
+    }
+
+    private bool adjust_delay() {
+        if (native != null) {
+            adjust_stream_delay(native);
+            return Source.CONTINUE;
+        } else {
+            adjust_delay_timeout_id = 0;
+            return Source.REMOVE;
+        }
+    }
+
+    private void process_outgoing_buffer(Buffer buffer) {
+        if (buffer.pts != uint64.MAX) {
+            last_reverse = buffer.pts;
+        }
+        analyze_reverse_stream(native, echo_probe.audio_info, buffer);
+        if (adjust_delay_timeout_id == 0 && echo_probe != null) {
+            adjust_delay_timeout_id = Timeout.add(5000, adjust_delay);
+        }
+    }
+
+    public override FlowReturn submit_input_buffer(bool is_discont, Buffer input) {
+        lock (adapter) {
+            if (is_discont) {
+                adapter.clear();
+            }
+            adapter.push(adjust_to_running_time(this, input));
+        }
+        return FlowReturn.OK;
+    }
+
+    public override FlowReturn generate_output(out Buffer output_buffer) {
+        lock (adapter) {
+            if (adapter.available() >= period_size) {
+                output_buffer = (Gst.Buffer) adapter.take_buffer(period_size).make_writable();
+                int old_gain_level = 0;
+                if (stream_volume != null) {
+                    old_gain_level = (int) (stream_volume.get_volume(Audio.StreamVolumeFormat.LINEAR) * 255.0);
+                    notify_gain_level(native, old_gain_level);
+                }
+                process_stream(native, audio_info, output_buffer);
+                if (stream_volume != null) {
+                    int new_gain_level = get_suggested_gain_level(native);
+                    if (old_gain_level != new_gain_level) {
+                        debug("Gain: %i -> %i", old_gain_level, new_gain_level);
+                        stream_volume.set_volume(Audio.StreamVolumeFormat.LINEAR, ((double)new_gain_level) / 255.0);
+                    }
+                }
+            }
+        }
+        return FlowReturn.OK;
+    }
+
+    public override bool stop() {
+        if (process_outgoing_buffer_handler_id != 0) {
+            echo_probe.disconnect(process_outgoing_buffer_handler_id);
+            process_outgoing_buffer_handler_id = 0;
+        }
+        if (adjust_delay_timeout_id != 0) {
+            Source.remove(adjust_delay_timeout_id);
+            adjust_delay_timeout_id = 0;
+        }
+        adapter.clear();
+        destroy_native(native);
+        native = null;
+        return true;
+    }
+}
\ No newline at end of file
diff --git a/plugins/rtp/src/voice_processor_native.cpp b/plugins/rtp/src/voice_processor_native.cpp
new file mode 100644
index 00000000..9b3292b8
--- /dev/null
+++ b/plugins/rtp/src/voice_processor_native.cpp
@@ -0,0 +1,141 @@
+#include <algorithm>
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+#include <webrtc/modules/audio_processing/include/audio_processing.h>
+#include <webrtc/modules/interface/module_common_types.h>
+#include <webrtc/system_wrappers/include/trace.h>
+
+#define SAMPLE_RATE 48000
+#define SAMPLE_CHANNELS 1
+
+struct _DinoPluginsRtpVoiceProcessorNative {
+    webrtc::AudioProcessing *apm;
+    gint stream_delay;
+};
+
+extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *transform, GstBuffer *buffer) {
+    GstBuffer *copy = gst_buffer_copy(buffer);
+    GST_BUFFER_PTS(copy) = gst_segment_to_running_time(&transform->segment, GST_FORMAT_TIME, GST_BUFFER_PTS(buffer));
+    return copy;
+}
+
+extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay) {
+    _DinoPluginsRtpVoiceProcessorNative *native = new _DinoPluginsRtpVoiceProcessorNative();
+    webrtc::Config config;
+    config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
+    config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85));
+    native->apm = webrtc::AudioProcessing::Create(config);
+    native->stream_delay = stream_delay;
+    return native;
+}
+
+extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::AudioProcessing *apm = native->apm;
+    webrtc::ProcessingConfig pconfig;
+    pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
+            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
+            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
+            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
+            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    apm->Initialize(pconfig);
+    apm->high_pass_filter()->Enable(true);
+    apm->echo_cancellation()->enable_drift_compensation(false);
+    apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression);
+    apm->echo_cancellation()->enable_delay_logging(true);
+    apm->echo_cancellation()->Enable(true);
+    apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate);
+    apm->noise_suppression()->Enable(true);
+    apm->gain_control()->set_analog_level_limits(0, 255);
+    apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
+    apm->gain_control()->set_target_level_dbfs(3);
+    apm->gain_control()->set_compression_gain_db(9);
+    apm->gain_control()->enable_limiter(true);
+    apm->gain_control()->Enable(true);
+    apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kLowLikelihood);
+    apm->voice_detection()->Enable(true);
+}
+
+extern "C" void
+dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    webrtc::AudioProcessing *apm = native->apm;
+
+    GstAudioBuffer audio_buffer;
+    gst_audio_buffer_map(&audio_buffer, info, buffer, GST_MAP_READ);
+
+    webrtc::AudioFrame frame;
+    frame.num_channels_ = info->channels;
+    frame.sample_rate_hz_ = info->rate;
+    frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf;
+    memcpy(frame.data_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf);
+
+    int err = apm->AnalyzeReverseStream(&frame);
+    if (err < 0) g_warning("ProcessReverseStream %i", err);
+
+    gst_audio_buffer_unmap(&audio_buffer);
+}
+
+extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::AudioProcessing *apm = native->apm;
+    apm->gain_control()->set_stream_analog_level(gain_level);
+}
+
+extern "C" gint dino_plugins_rtp_voice_processor_get_suggested_gain_level(void *native_ptr) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::AudioProcessing *apm = native->apm;
+    return apm->gain_control()->stream_analog_level();
+}
+
+extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *native_ptr) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::AudioProcessing *apm = native->apm;
+    return apm->voice_detection()->stream_has_voice();
+}
+
+extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::AudioProcessing *apm = native->apm;
+    int median, std;
+    float fraction_poor_delays;
+    apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays);
+    if (fraction_poor_delays < 0) return;
+    g_debug("voice_processor_native.cpp: Stream delay metrics: %i %i %f", median, std, fraction_poor_delays);
+    if (fraction_poor_delays > 0.5) {
+        native->stream_delay = std::max(0, native->stream_delay + std::min(-10, std::max(median, 10)));
+        g_debug("Adjusted stream delay %i", native->stream_delay);
+    }
+}
+
+extern "C" void
+dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
+    webrtc::AudioProcessing *apm = native->apm;
+
+    GstAudioBuffer audio_buffer;
+    gst_audio_buffer_map(&audio_buffer, info, buffer, GST_MAP_READWRITE);
+
+    webrtc::AudioFrame frame;
+    frame.num_channels_ = info->channels;
+    frame.sample_rate_hz_ = info->rate;
+    frame.samples_per_channel_ = info->rate / 100;
+    memcpy(frame.data_, audio_buffer.planes[0], frame.samples_per_channel_ * info->bpf);
+
+    apm->set_stream_delay_ms(native->stream_delay);
+    int err = apm->ProcessStream(&frame);
+    if (err >= 0) memcpy(audio_buffer.planes[0], frame.data_, frame.samples_per_channel_ * info->bpf);
+    if (err < 0) g_warning("ProcessStream %i", err);
+
+    gst_audio_buffer_unmap(&audio_buffer);
+}
+
+extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) {
+    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
+    delete native;
+}
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